Portsip PBX User Guide v12.8 en

PortSIP PBX User Guide
Version: v12.8.3
Date: March 8, 2023
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Summary of Changes
Changes for Release v12.8.3
The following changes are included in this release:
• To disable the PBX from sending a welcome email, set the extension custom
options to { "disable_welcome_email" : true}.
• The "extension_presence" event has been added for when an extension changes
its - presence status.
• The "cdr_id" field has been added to the CDR event.
• Fixed an issue where the CDR was incorrect if the caller or callee registered to the
PBX from multiple devices and launched a call via REST API.
• Fixed an issue where intercom/paging calls would automatically hang up after 60
seconds.
• Fixed an issue where an agent who signed out of the PBX and then signed back in
would have a “Ready” status but could not receive distributed calls.
• Fix the RTP packet issue with the marker bit flag that is incompatible with some
trunks.
• Fix the issue where there is no ringback tone during IVR forward calls.
• Fix the issue if there are many holidays, none of them are affected.
• Fix the issue if the holiday falls on a different year.

• Fix a potential crash bug of the WebRTC call

• Correct the security issue with the web portal server settings.
• Fix the issue might lost the recorded file
• Fix the issue if the call is forwarded from the IVR/Queue/Ring Group and the
recording option is only enabled for the extension, the call will not be recorded.
• Large improvements in performance

• Fix a potential crash bug if the HA is enabled

• Improvements in performance

• Allows any extension obtain the contacts from PBX


• Rewrite the push feature to improve the stability

• Fix up if reinstall the mobile app cause the app keep receiving push notificaitons
• Provide support for provision line key configuration to IP Phone
• Improve IP Phone re-provisioning; the IP Phone does not need to be rebooted to
apply new settings

• OpenSSL was updated to 1.1.1n to avoid CVE-2022-0778

• Resolve an issue where the SIP scanner generates ghost calls (calls that cannot
be established but generate spam CDR)
• Fix an issue that might cause the media server to crash if there were a large
number of calls on the PBX for a lengthy period of time - a few weeks
• Fix a bug that in some scenarios let the holiday appear as an office hour
• By default, the Flow-Timer(RFC5626) header is disabled
• Fix the CANCEL reason header format issue
• Before the delete operation, a message box pops up for the user to confirm

• Support the tel scheme

• Support mobile push notifications for the intercom group
• Replace the host IP in the Via header with the PBX public IP when the PBX sends
a request to the trunk
• Corrected a problem that prevented multiple Timer tokens from being added to the
Supported header
• Fixed the format of the Reason header
• If a mobile app hasn't been registered with the PBX within a week, clear the push
information
• Fix an issue in which if the client register to PBX over TCP/TLS/WSS, the call
would be hung up after 9-10 minutes
• Fixed a bug if the yealink phone performs an attended transfer, the call will not be
hangup
• Fix a bug when send massive mobile push notifications cause PBX crashed

• When the PBX detects that the WSS/TLS/TCP client connection is down, then
clean the established calls
• Fix a bug where there is no voice when a call is transferred to a mobile app that
has enabled push notifications
• Fix a bug in the BYE message where the reason header format is wrong

• SSv2, v3, TLS 1.0, 1.1 have been disabled

• For the blind transfer and picked ringing call, combine the CDR and recording file
into a single record
• Add a custom option "no_external_recording" to stop call recording when a call
is made between two external numbers.
• Add a custom option called "www-authentication" that allows the PBX to use the
www-authentication mechanism
• The call hold status is no longer displayed on the Web Portal
• For Android, fix the crash bug with push notifications
• In the HA deployment, fix the bug where the WebRTC client has no voice
• The WebSocket Interface (WSI) now allows a subscriber to subscribe from multiple
locations
• When a queue/group member receives a call from a ring group/queue, add the
queue number/group number to the Remote-Party-ID and P-Asserted-Identity
headers
• Enhance High Availability, voice will no longer be interrupted if the master server is
unavailable
• The recording files will be deleted after 180 days by default
• Fix the bug the caller will no longer hear the MOH if the callee repeats hold/unhold
the call a few times
• Fix the bug of /api/comm_message/update and /api/comm_message/list
• Fix the bug that for long time running, the PBX stopped to send the email
notification

• Support capture the SIP message and analyze it for troubleshooting on the Web
UI with PortSIP Trace Server
• Force the extension and tenant use the strong password
• When an admin/tenant/extension changes his password, must enter the current
password.
• Add the queue_member_state message event of topic QUEUE_EVENTS for the
WebSocket Publisher, when an agent of the call queue changed his state to ready/
not-ready by REST API or dial code, this event will be pushed.
• Password settings in the Web Portal have been moved to a menu that appears
when you click on the avatar.
• Remove the log file path from the log file list view
• Allow users to set a wildcard * for the "Exceptions" of extension forwarding rules,
allowing them to designate a break time during business hours
• Fix the BASIC authorization bug for CDR events and Extension events
• Fix the crash bug that occurs when two extensions set forward calls to each other
when one of them is busy
• Fix the bug if caller make call to extension from PSTN, and caller hold the call, the
MOH is not affected
• Set the minimize of registration time to 300 seconds
• Add the parameter "member_number" of REST API "api/call_queues/
member_state/update" to instead of the parameter extension_id
• Improve the WebRTC client and Windows client App

• Removed the built-in backup/restore features, suggest use the VM snapshot to

backup. Or user can simply backup the data directory:
a. Windows: by default it's
c:/programdata/portsip
b. Linux: by default it's

/var/lib/portsip
• Support store the recording and log files to the AWS S3.
• Support the Privacy header, if the UA makes call with the Privacy header, the PBX
will forward this header to the callee.
• When there have the call reached agent of the ring group/queue, the event will be
published to the WebSocket Publish subscriber.
• Provide a custom option allows includes the recording file ID in the CDR event
JSON payload.
• The REST API for list the CDR, call recording, black list is changed.
• Fixed a crash bug if works with the MicroSIP App for presence.
Chapter 1. Getting Started with PortSIP® PBX

This guide is designed to assist administrators deploying PortSIP products in a Windows or
Linux environment, and explain a number of deployment modes, architectures, and limitations
of the solution.
1.1 What is PortSIP PBX

PortSIP PBX (also known as PortPBX, PortSIP UC Server) is a software-based Unified
Communications system for Windows and Linux that works with SIP standard-based IP
Phones, Softphones, SIP Trunks and VoIP Gateways to provide a complete PBX solution –
without the inflated cost and management headaches of an "antiquated" PBX. The SIP PBX
supports not only all traditional PBX features, but also includes many new mobility and
productivity features.
Calls are sent as data packets over the computer data network instead of the traditional phone
network. Phones share the network with computers so no separate phone wiring is required.
With the use of a VoIP Provider, SIP Trunk, you can connect existing phone lines to the
PortSIP PBX to make and receive phone calls via a regular PSTN line. You can also use a
VoIP Provider, which removes the requirement for a gateway. PortSIP PBX interoperates with
standard SIP softphones, IP phones or smartphones, and provides internal call switching.
1.2 Before Started

Prerequisite knowledge for Linux
Deploying PortSIP PBX in a Linux environment requires planning and knowledge of session
initiation protocol (SIP) audio, video call and presence, Instant Messaging (IM) administration.
You should also have knowledge of the following Linux infrastructures:
A popular Linux distribution:
• CentOS 7.9 (64-bit)

• Debian 10.x (64-bit)
• Ubuntu 18.04 or 20.04 (64-bit)
• Docker 20.10 or higher
• IPv4/IPv6
• Systemd
• IP tables
• Firewalld
• UFW
This document assumes that the Linux OS is already deployed and administrators of PortSIP
PBX have been allocated with the root permission to Linux.
Prerequisite knowledge for Windows

Deploying PortSIP PBX in a Windows environment requires planning and knowledge of
session initiation protocol (SIP) audio, video call and presence, Instant Messaging (IM)
administration. You should also have knowledge of the following Windows infrastructures:
A Windows desktop or Windows server OS:
• Windows 10 (64-bit)
• Windows Server 2012 R2
• Windows Server 2016 R2 or higher
• IPv4/IPv6
• Windows firewall
This document assumes that the Windows OS is already deployed and administrators of
PortSIP PBX have been allocated with the administrator permission to Windows.
Cloud and Virtualization Environment Supported

To build high-availability communication solution to help clients reducing cost and improving
communication performance, PortSIP PBX commits support on cloud services and have
confirmed compatibility with following cloud and virtualized environment:
• VMware ESX 5.X and above.

• Linux HyperV
• Microsoft HyperV 20012 R2 and above
• Microsoft AZURE
• Amazon AWS
• Google Could
• Digital Ocean
• UCloud
System performance depends on following key factors:
• Maximum simultaneous calls needed for PBX

• Maximum online users needed for PBX
• Recordings for calls
• Recording audio only or both of audio and video
• Maximum online users for audio/video conferences on PBX
• Maximum IVRs (Virtual Receptionist) on PBX
• Maximum Call Queues on PBX
• Maximum Ring Groups on PBX
Depending on the key features listed above, PortSIP PBX is able to run on PCs and servers
with various CPUs ranging from Intel i3 CPUs to Inter Xeon.
Other Requirements
• Latest Firefox, Google Chrome, Edge browser
• Microsoft .NET Framework version 4.5 or higher
• Knowledge of Linux and Linux Internet administration
• Knowledge of Windows and Windows Internet administration
• A constant Internet connection to stun4.l.google.com on port 19302
• Ensure server date time is synced correctly.
FQDN Support
Although PortSIP PBX is designed to be able to run on servers without FQDN specified, we
recommend to specify FQGN with following advantages:
• Easier access to Web Portal for PortSIP PBX

• Easier management of IP phones and clients after IP address change for PBX
• Convenient access to HTTPS when accessing Web Portal
• Avoid browser warning when access the WebRTC Client
The FQDN you are using must be able to be resolved correctly into the server with PortSIP
PBX installed in LAN. If PortSIP PBX is installed on public network, FQDN must be resolved
correctly into the public network address for server with PBX installed.
1.3 Getting Help and Support Resources

Official PortSIP Support Community
PortSIP Knowledge Base
Submit a request
Email support
Chapter 2. Installation of PortSIP® PBX

This chapter provides the instructions for installing the PortSIP PBX on Windows and Linux.
2.1 Downloading PortSIP PBX

The latest free edition of PortSIP PBX could always be found and downloaded at PortSIP
Website. It’s available for both 64-bit Windows and Linux, but not for 32-bit version.
The free edition of PortSIP PBX offers a maximum of 3 simultaneous calls and 10 extension
registrations. If you require more simultaneous calls/extensions, please refer to [License
Section](#11.10 License) for more details.
You will get the installer after download completed.
2.2 Installing PortSIP PBX on Linux

PortSIP PBX Linux edition is migrated to docker environment, which does not support RPM
and Deb installer.
The OS required:
• CentOS: 7.9
• Ubuntu: 18.04, 20.04
• Debian: 10.x
• Only supports 64bit OS
Important
From v12.6.1, the PortSIP PBX requires running with the above Linux OS versions. If there
already installed the PortSIP PBX which less than v12.6.1, and wish to upgrade to v12.6.1 or a
later version, must upgrade the Linux OS to the above version before upgrade the PortSIP
PBX.
Preparing the Linux Host Machine for Installation

Tasks that MUST be completed before installing PortSIP PBX:
• If the Linux on which PBX will be installed is located in LAN, assign a static LAN IP
address; if it's in public network, please assign static IP address for public network
• Install all available updates & service packs before installing PortSIP PBX
• Do not install VPN software on your PortSIP PBX Server
• Do not install PostgreSQL on your PortSIP PBX Server
• Ensure that all power saving options for your System and Network adapters are
disabled (by setting the system to High Performance)
• Do not install TeamViewer, VPN and other similar software on the host machine
• PortSIP PBX must not be installed on a host which is a DNS or DHCP server
• Below ports must be permitted by your firewall.
UDP: 45000– 65000, 25000- 35000
TCP: 8899– 8900、8887-8888、8881-8885
• Make sure that below ports have not been used by other programs:
UDP: 45000– 65000, 25000- 35000
TCP: 8899– 8900、8887-8888、8881-8885
IMPORTANT: If you running the PBX on the cloud platform such as AWS, and the
cloud platform has the firewall itself, you MUST open the ports on the cloud
platform firewall too.
Installing a fresh PortSIP PBX v12.x for Linux

To install the PortSIP PBX for Linux, please refer to: Setup PortSIP PBX for Linux
Configuring Linux Firewall Rules

After having successfully installed PortSIP PBX, the PortSIP PBX ports has been opened with
Linux firewall.
If your server has a firewall which is blocking the ports, you must open the below ports in order
to make the PortSIP PBX working properly.
• UDP ports: 45000– 65000, 25000- 35000. These ports are used for the RTP
sessions.
• TCP: 8899– 8900、
、8887-8888、
、8881-8885. These ports are used for the Server
control and WebRTC client.
• UDP: 5060. This is the default UDP transport for SIP communications (to send and
receive SIP signaling).
• TCP: 5065. This is the default WSS transport for SIP communications in browser
(to send and receive SIP signaling).
You also need to open the port that you are using for adding new transport:
• Assume you have added a TLS transport on port 5063, you must open TCP port
5063 in your Linux firewall and
• Assume you have added a TCP transport on port 5061, you must open TCP port
5061 in your Linux firewall
• Assume you have added a UDP transport on port 5068, you must open UDP port
IMPORTANT: If you running the PBX on the cloud platform such as AWS, and the cloud
platform has the firewall itself, you MUST open the ports on the cloud platform firewall too.
2.3 Installing PortSIP PBX on Windows

Preparing the Windows Host Machine for Installation
Tasks that MUST be completed before installing PortSIP PBX.
• If the Windows PC / server on which PBX will be installed is located in LAN, assign
a static LAN IP address; if it's in public network, assign a static IP address for
public network.
• Install all available Windows updates & service packs before installing PortSIP
PBX. The reboot after installing Windows updates may reveal additional updates.
Pay particular attention to install all updates for Microsoft .Net before running the
PortSIP PBX installation.
• Anti-virus Software should not scan the following directories to avoid complications
and write access delays: C:\Program Files\PortSIP; C:\Programdata\PortSIP
• Do not install VPN, TeamViewer software on your PortSIP PBX Server
• Do not install PostgreSQL on your PortSIP PBX Server
• Ensure the “Windows Firewall” service has been started.
• Ensure that all power saving options for your System and Network adapters are
disabled (by setting the system to High Performance).
• Disable Bluetooth adapters if it is a Windows client PC.
• PortSIP PBX must not be installed on a host which is a DNS or DHCP server, or
that has MS SharePoint or Exchange services installed.
• Below ports must be permitted by your firewall:
UDP: 45000– 65000, 25000- 35000
TCP: 8899– 8900、8887-8888、8881-8885
• Make sure that below ports have not been used by other programs:
UDP: 45000– 65000, 25000- 35000
TCP: 8899– 8900、8887-8888、8881-8885
• Ensure your Windows Firewall is enabled
Installing a fresh PortSIP PBX v12.x for Windows
To install PortSIP PBX, you only need to double-click the installer, which will guide you through
the installation process.
PortSIP PBX services will automatically start after successful installation (and there after every
time your computer starts up).
Important: during installation, when you choose the folders for the PBX, below two
folders must not same!!
Configuring Windows Firewall Rules

After having successfully installed PortSIP PBX, the PortSIP PBX ports has been opened with
Linux firewall.
If your server has a firewall which is blocking the ports, you must open the below ports in order
to make the PortSIP PBX working properly.
• UDP ports: 45000– 65000, 25000- 35000. These ports are used for the RTP
sessions.
• TCP: 8899– 8900、
、8887-8888、
、8881-8885. These ports are used for the Server
control and WebRTC client.
• UDP: 5060. This is the default UDP transport for SIP communications (to send and
receive SIP signaling).
• TCP: 5065. This is the default WSS transport for SIP communications in browser
(to send and receive SIP signaling).
You also need to open the port that you are using for adding new transport:
• Assume you have added a TLS transport on port 5063, you must open TCP port
5063 in your Linux firewall and
• Assume you have added a TCP transport on port 5061, you must open TCP port
• Assume you have added a UDP transport on port 5068, you must open UDP port
2.4 Upgrade the PortSIP PBX

Please follow this link to upgrade the PortSIP PBX: Upgrade PortSIP PBX
2.6 Avoiding HTTPS Certificate Security Warnings

PortSIP PBX listens on 8888 port for providing HTTP portal to access the PBX Web Portal.
Assume your server IP is 172.217.14.16, you should open this URL: http://172.217.14.16:8888
by your browser. Note: Chrome and Firefox is recommended, please don’t use IE.
PortSIP PBX listens on 8887 port for providing HTTPS portal to access the PBX Web Portal.
Assume your server IP is 172.217.14.16, you should open this URL:

https://172.217.14.16:8887 by your browser. Note: Chrome and Firefox is recommended.
Please do not use IE.
For HTTPS portal default usage of the self-signed SSL certificate will cause the browser
popups SSL certificate security warning.
To avoid SSL certificate warning, you will need to purchase a Signed Certificate (which is an
authorized certificate issued by trustworthy certificate authority) to replace the self-signed one.
To do this, please:
• Resolve your PBX web domain (for example mypbx.com, if you don't have the
domain you can purchase it from domain provider, such as Godaddy) to the PBX
IP in case is 172.217.14.16.
• Go to Thawte or Versign or other certificate providers to purchase a SSL certificate
for your PBX Web Domain(in case is mypbx.com). Save the private key as
portsip.key
• After you have obtained the SSL certificate, rename the certificate to portsip.crt
Now place the portsip.crt and portsip.key files in a folder we will use it later.
Note: You may also obtain SSL certificate from Let’s Encrypt for free.
Chapter 3. Architecture and deployment of

PortSIP® PBX
PortSIP PBX consists of various components, including Call Manager, Media Server, REST
API Gateway, File Server, WebRTC Gateway, Database, Call Queue, Conference, Voicemail,
MOH, Auto Attendant, Instant Messaging, and Presence Server. Below is a high level
architectural diagram about how it works.
PortSIP PBX could be deployed in wide range of scenarios. It is supported in LAN and
Internet, as well as popular cloud platforms, such as AZURE, AWS, Digital Ocean and Google
Cloud, UCloud, Tencent Cloud, Alibba Cloud.
After successful installation of PortSIP PBX with setup wizard, all you need is a few clicks to
make it running.
Running the PortSIP PBX Configuration Wizard

The PortSIP PBX configuration wizard will guide you through a number of essential tasks to
get your system up and running.
PortSIP PBX listens on 8888 port for providing HTTP portal to access the PBX Web Portal,
and listens on 8887 port for providing HTTPS portal. More details please read the [2.6
section](#2.6 Avoid HTTPS Certificate Security Warnings).
• Access the PBX Web Portal by visiting http://172.217.14.16:8888 or

http://mypbx.com:8888 if you resolved your PBX web domain to PBX IP.
• Enter the Username and Password (defaulted as ”admin” for both) and click the
“Sign in” button. Note that both the Username and Password are case sensitive.
The “Setup Wizard” will be displayed which will guide you through the initial
configuration step by step
You may change the default Username and Password for "admin" by navigating to "Profile" >
"General" in PortSIP PBX Web Portal.
3.1 Deploying PortSIP PBX in LAN

Assuming that PortSIP PBX is deployed in LAN with Internet connection, the server/PC has
installed the PBX and the private IP is 192.168.0.16. As the PBX is connected to SIP trunk or
VoIP provider, users can not only make & receive calls in LAN, but also make & receive
external calls with PSTN number and mobile users via services provided by pre-configured SIP
trunk or VoIP providers.
Step 1:
If you want to use the HTTPS with PortSIP PBX Web Portal and WebRTC client,, you must
set up the "Web Domain" here, and prepare a SSL certificate for this "Web Domain" since the
browser is requires a trusted certificates otherwise it will block the HTTPS and WebRTC
Client. In case we use the mypbx.com. You will also need resolve your Web domain
mypbx.comto your PBX server IP.
Enter the Public IPv4 if you have a static public IP of your LAN. Do not enter the Public IPv4 if
your public IP is dynamic.
Note: the loopback interface (127.0.0.1) is unacceptable. Only the static IP for LAN where
the PBX is located is allowed (do not use DHCP dynamic IP). This private IP must be
reachable by your SIP client.
The IP address entered here is the SIP server IP address for PBX. It is required when a SIP
client or SIP IP phone registers to PortSIP PBX should be configured as the "Outbound Proxy
Server".
Step 2:
You will now need to enter your SIP domain here. The SIP domain is usually a FQDN (Full
Qualified Domain Name). You could use IP address instead if you don’t have an FQDN. The
SIP domain does not have to be resolvable, it’s used for PBX authentication purpose only.
After setting up the SIP domain (in this case it is portsip.io), the extension SIP URI will be
sip:xxx@portsip.io. For example, the extension 101 SIP address would be: sip:101@portsip.io.
If you don't want to use domain here, enter the private IP (for example: 192.168.0.16) of the
Server which has installed the PortSIP PBX instead of the domain (FQDN). In this case the
extension 101 SIP address would be: sip:101@192.168.0.16.
Step 3:
You can set transport layer protocol for the SIP signaling here, with the default transport UDP
on port 5060.
Note: You can add more transports in PortSIP PBX Web Portal after this Wizard.
Step 4:
If you want to use the HTTPS with PortSIP PBX Web Portal and WebRTC client, you must
upload the SSL certificates file here for WSS transport, by default the PBX listens 5065 port
for WSS transport which communicates with the WebRTC client.
Note: You can use the self-signed certificates here but it will cause browser pop ups the
warning when you open the WebRTC client, you can purchase a trusted certificate to avoid the
browser blocking and warning. More details please read the [2.6 section](#2.6 Avoid HTTPS
Certificate Security Warnings).
Step 5:
Setup mail server. You may set up the SMTP mail server in this step for receiving notifications,
voicemails, conference invitations and CDR downloads. You can use your SMTP server or
Gmail SMTP server.
Note: This step is not mandatory. You may choose to setup SMTP server whenever necessary.
By clicking the “Apply” button, you have now completed the initial configuration of PortSIP
PBX. You will be redirected to Web Portal.
Step 6
After successfully completed the setup wizard, you have to restart the PortSIP PBX in order to
make the SSL certificates work.
Linux:
$ sudo docker exec -it portsip-pbx /bin/bash

$ supervisorctl stop nginx
$ supervisorctl stop gateway
$ supervisorctl start nginx
$ supervisorctl start gateway
Windows:
Restart the Windows Server directly.
After restarted, you can sign in PortSIP PBX Web Portal by URL https://mypbx.com:8887
If you don't use trusted certificate files for the WSS transport, you will get the browser warning
and blocked when you use WebRTC client.
Please refer to [2.6 Avoid HTTPS Certificate Security Warnings](#2.6 Avoid HTTPS Certificate
Security Warnings) to learn how to avoid the browser warning and blocking.
3.2 Deploying PortSIP PBX in the cloud

PortSIP PBX can be deployed on popular cloud platforms such as AWS, GCE, AZURE, Digital
Ocean.
In this section we will use the AWS as an example, as the other cloud platforms are quite
similar to the way how AWS works.
When PortSIP PBX is deployed on AWS, user can make or answer calls through PortSIP PBX
with other users through Internet, and make or answer external calls via SIP trunk or VoIP
provider.
Please refer to Creating an AWS account if you do not have an AWS account, and you will
need to launch an EC2 instance for installing the PortSIP PBX.
Step 1:
On the left bar of AWS EC2 Web Portal, choose “Elastic IPs”. As you see the “Elastic IP”,
please write it down for future use. If the “Elastic IPs” does not exist, click “Allocate New
Address”, and associate the Elastic IP to your instance.
Step 2:
In the Configuration Wizard of PortSIP PBX, enter the private IP of AWS EC2 as "Private
IPv4", and enter the “Elastic IP” of AWS EC2 as "Public IPv4".
Now remaining steps are same to the "[3.1 Deploy PortSIP PBX in LAN](#3.1 Deploy PortSIP
PBX in LAN)"
3.3 Deploying PortSIP PBX in other scenarios

If you would like to deploy PortSIP PBX in other scenarios which are not mentioned above, you
will need to get the server IP address (public IP and private IP) and enter in the PBX setup
wizard and follow the Configuration Wizard for deployment.
Note: if the server only has public IP but no private IP, please enter the public IP for both
Private IPv4 and Public IPv4.
Chapter 4. PBX Overview
4.1 Summary
In this page we can have an overview of the PBX statistics:
• Version
• Uptime
• Events
• Extensions
• Tenants
• Active Calls
• CPU and memory usage
• Call statistics for the past 24 hours
4.2 Services Status

services
You may navigate to “Home > Service Status” menu in the PortSIP PBX System Web Portal
to quickly view if all PortSIP PBX system services are working correctly.
4.3 Network
This section provides an overview of the general network configurations of PBX, including DNS
Server, PBX IP, SIP domain, transports. These settings are useful for SIP clients to register to
PBX.
4.4 System extensions

PortSIP PBX defines services such as Virtual Receptionist (auto attendant), Conferencing,
Ring Group, Call Queue, voicemail and Music on Hold as system extensions, which could be
used by PBX only. To check if the system services are correctly registered to PBX, please go
to “Home > System Extensions” menu in the PortSIP PBX System Web Portal.
Chapter 5. Call Manager

After completing the Configuration Wizard, you may manage PortSIP PBX in the Web Portal.
5.1 Domain and Transport

The SIP domain is used during client registration and calling, and it should match the domain
part of your own SIP address on your phone - i.e. if other people are going to call your phone,
they must use that domain name as part of the SIP address they use to reach you. The
domain can be a FQDN or the IP address, for example “portsip.io” or “192.168.0.16”.
The SIP domain is configured within “Setup Wizard” when you sign into Web Portal for the first
time. To modify a SIP domain, go to “Call Manager > Domain and Transport”, and click “Edit”
button to enter new SIP domain and save.
PortSIP PBX supports a wide range of transports, including UDP, TCP, TLS, WSS (WebSocket
Security) for SIP message. You need to configure the transport, and set the ports to use when
listening for SIP messages.
The default transport has been configured with “Setup Wizard”. To make changes, you need
to select the “Call Manager > Domain and Transport” menu, and click “Add” button in
“Transports” section. The domain must be added before you add a new transport.
Note: only administrator are allowed to create or delete SIP transport. When deleting, at least
one transport must be left around.
Add UDP/TCP/WS transport

To add UDP/TCP/WS transport:
• Click the “Add” button, choose the UDP/TCP/WS in “Transport protocol” box.
The default Transport Port for UDP/TCP/WS is 5060/5063/5062. You may specify
another port as you like, but the port must not be in use by other applications
• Click the “Apply” button to add the transport
Add TLS/WSS transport

In order to use the WebRTC Client, we must add the TLS/WSS transport with self-signed
certificate.
If you already uploaded the SSL certificate files in the step 4 of "Section 3.1", then the WSS
transport is already added.
First of all, prepare the certificate files.
• You have to generate the certificate files by yourself if you have not purchased
trusted certificates from a trusted third-party certificate provider(for example,
versign, thawte, digicert). Please download the certificate file tool from PortSIP
website (or run PortCertMaker.exe in the installation path of PBX), enter your
"PBX Web domain" which you entered in the step 1 of "Setup Wizard". Once
clicked “Generate” button, certificate files will be generated automatically.
• The certificates include three files (assume your "Web Domain" is mypbx.com):
domain_key_mypbx.com.pem
domain_cert_mypbx.com.pem
root_cert_mypbx.com.pem
• Rename domain_key_mypbx.com.pem to portsip.key, rename
domain_cert_mypbx.com.pem to portsip.crt
You can also follow below steps if you would like to purchase certificate files from a trusted
third-party provider (assume purchased certificate for mypbx.com):
• Generate the CSR file and private key file according to provider’s guide, and keep
the files. If you have set the password when generating the private key file,
remember it for future use;
◦ Rename the private key file as portsip.key
◦ Submit the CRS file to provider, and download the certificate files after
your certificates approved. This step will end up with two files:
Intermediate CA certificate and SSL certificate
◦ Use a plain text editor for example Windows Notepad (do not use MS
Word) to open the Intermediate CA file and SSL certificate file, copy
the Intermediate CA contents to append to the SSL certificate file, and
rename SSL certificate file as portsip.crt.
• Click “Add” button and choose the TLS or WSS in “Transport protocol” box. The
default Transport Port for TLS/WSS 5063/5065. You may specify another port as
you like, but the port must not be in use by other applications
• Click the Upload button to choose the certificate files that you have generated for
uploading, “portsip.crt” for the “Certificate file”, “portsip.key” for the “Private
key file”.
• Click the “Apply” button to add the transport
Important: after added TLS/WSS transport，

，please perform below commands to
restart server.
Linux:

Windows:
Restart the Windows Server directly.
After restarted, you can sign in PortSIP PBX Web Portal by URL https://mypbx.com:8887
Please refer to [2.6 Avoid HTTPS Certificate Security Warnings](#2.6 Avoid HTTPS Certificate
Security Warnings) to learn how to avoid the browser warning and blocking.
Firewall for new added transports

You have to edit your firewall rules to permit the port that you specified for the transports. For
example, you have added below transports in PortSIP PBX:
UDP: 5060
TCP: 5061
TLS: 5063
WS: 5064
WSS: 5065
Then you must add below firewall rules for your PortSIP PBX:
UDP: 5060 from IP: 0.0.0.0(anywhere)
TCP: 5061 from IP: 0.0.0.0(anywhere)
TLS: 5063 from IP: 0.0.0.0(anywhere)
5.2 Phones
Phone Auto Provisioning
Once PortSIP PBX is installed, you can configure your IP phones and assign an extension to
each phone.
Phones can be configured one by one manually using their web interface, which is time
consuming and leads to many errors; Or by using phone provisioning feature offered by
PortSIP PBX, which makes it possible to manage phones centrally and remotely and without
having to login to the phone’s web interface one by one. Using this method you instruct the
phone to retrieve its configuration from PortSIP PBX.
Phone provisioning greatly eases day to day management of IP phones. It makes it easy to
change extension passwords, BLF lights and so on because you can do it centrally for all
phones from the PortSIP PBX Web Portal and then push the changes to the phone. The
following provisioning methods are supported:
• Plug and Play - Supported IP phones can be provisioned automatically using plug
and play (Applicable for phones on the local LAN)
• Via RPS - For the Yealink, Fanvil, Htek IP Phones, they can be provisioned by the
RPS
• Via Manual Provisioning URL - Supported IP phones can be provisioned by
entering the provisioning URL into the phone's web interface (Applicable for local,
remote and SBC extensions)
• Via DHCP Option 66 - Legacy phones (from a previous PBX installation, e.g.
Polycom, Cisco or Aastra) can be provisioned via DHCP for use in the local LAN
only. Some limitations apply
You can find a list of supported and legacy phones here. Additional half an hour to provision
the phones saves more hours from future efforts!
Provisioning phones by using Plug and Play (for local LAN)
Note: PnP provisioning requires that the PortSIP PBX runs on the default sip port 5060 and
that the IP phones resides on the same local LAN subnet as PortSIP PBX.
To auto provision phones using Plug and Play:
1. Plug the phone into the network

2. The phone will send a multicast message across the LAN, this will be picked up by
PortSIP PBX
3. The phone will show up in the “Call Manager > Phones” menu in the PortSIP PBX
Web Portal as a new phone
4. Assign the phone to an existing extension or create a new one for the phone
5. Go to the extension’s “Phone Provisioning” tab and specify other configuration
settings for the phone
6. Enter the password for access IP Phone web UI
7. Select “Phone Display Language” and “Timezone” for the phone
8. Click “OK”
9. The PBX send the configuration file URL to IP Phone, IP phone will download the
configuration file
10. The phone will apply the settings and connect to PortSIP PBX. The IP phone will
be manageable from within the PortSIP PBX Web Portal
Provisioning Phones using provisioning link manually

Remote phones that are not in same LAN with PortSIP PBX, it must be configured manually by
the provisioning link. To provision a remote phone:
1. From the “Call Manager > Phones” menu in the PortSIP PBX Web Portal, select
“Add Phone”.
2. Select the extension that the phone uses.
3. Enter the MAC address of the phone (which can be found at the bottom of the
phone).
4. Select the appropriate phone model from the drop down menu.
5. Select “Phone Display Language” and “Timezone” for the phone.
7. Click "Apply" button and then edit this extension.
8. Copy the provisioning link.
9. Insert the provisioning link manually into the phones. You can find it in “Phone
Provisioning” tab of extension configuration.
Provisioning remote Phones by RPS
In the case PortSIP PBX locates in cloud, IP Phones can be auto provisioned via RPS (RPS is
a service provided by IP Phone vendors). PortSIP PBX supports for Yealink, Htek, and Fanvil
RPS. If you are using the IP Phone from one of the providers above, you may configure your
IP Phone in a very easy way, without the need to manually copy the auto provisioning link (in
this scenario the PnP mode is unavailable).
1. From the “Call Manager > Phones” menu in the PortSIP PBX Web Portal, select
“Add Phone”.
2. Select the extension that the phone uses.
3. Enter the MAC address of the phone (which can be found at the bottom of the
phone).
4. Select the appropriate phone model from the drop down menu.
5. Select “Phone Display Language” and “Timezone” for the phone.
7. Click "Apply" button, the PortSIP PBX will writes the provisioning link to the IP
Phone's RPS. Once the IP Phone is started, it will query the provisioning link with
its MAC address to complete the auto provisioning.
Provisioning Legacy phones: Cisco, Polycom & Aastra
Cisco, Polycom and Aastra phones do not support plug and play nor secure HTTPS
provisioning with a Let’s encrypt Root CA or self-signed CA. They can only be used on the
local LAN and must be provisioned as follows:
1. Download the firmware that has been tested by PortSIP with the legacy phones.
2. Factory reset your phones to ensure that there are no old settings that might
conflict with the new configuration. Find out how here for Aastra, Cisco, Cisco SPA
and Polycom SoundPoint / SoundStation.
3. Now add the phone to an extension. You can do this from the phones page or you
can go straight to the extension, provisioning tab. Click “Add Phone”.
4. Select your phone model
5. Enter the MAC address of the phone. You will be taken to the provisioning page
6. Select “Phone Display Language” and “Timezone” for the phone
8. Click “Apply” to add the phone to the extension
9. IMPORTANT: Please take note of the provisioning link shown on “Auto
Provisioning” tab.
Now configure the phone to retrieve the configuration from the PortSIP provisioning folder. Use
DHCP option 66 or configure the phones manually via their web interface with the PortSIP
provisioning link. Cisco 7940/7960 phones must be provisioned using TFTP and DHCP option
66.
Detailed Step by Step guides for legacy phones:

Provisioning Polycom IP Phones
Provisioning Cisco 7940/ 7941/ 7960 /7961 phones
Provisioning Cisco SPA 302, 303,501G, 502G, 504G, 508G, 509G, 525G/G2
Provisioning Aastra 6730i, 6731i, 6739i, 6751i, 6753i, 6755i, 6757i
See Also
Remote phones? Read our guide on Provisioning a Remote Extension.
Using Provisioning IP Phone via DHCP 66 to configure the provisioning URL for legacy
phones.
See the list of Supported IP Phones by PortSIP PBX.
Setting up a TFTP server for firmware updates.
Factory resetting Aastra, Cisco, Cisco SPA, Gigaset, Panasonic, Polycom SoundPoint,
Polycom Soundstation, Yealink.
PnP auto provision IP Phone Multicast Debug
5.3 Managing Phones

PortSIP PBX provides an easy way to monitor and manage your phones and softphones
throughout your network. The “Call manager > Phones” menu in the PortSIP PBX Web Portal
allows you to:
• View all the phones in the network, including IP and MAC.

• View all PortSIP Clients connected in softphone mode.
• Check the firmware version that the phone is running.
• Remotely reboot one or all of the phones.
• Re-provision the phones.
• Launch the admin interface of the phone.
• Monitor security of extension password and PIN. Weak extension passwords and
PINs are the most common cause of security breaches.
Adding Phones
You can add phones to PortSIP PBX in the following ways:
• Plug and Play - Plug in the phone in the local LAN

• By MAC - for legacy phones
• By RPS - for remote phones
Plug and Play (LAN & SBC)

If you are connecting a supported phone that is on the same LAN as PortSIP PBX, you will see
the phone appear on the phones page, with its entry in BOLD. This means PortSIP PBX has
detected a new phone on the network that you need to process.
Select the phone and decide to:
1. Assign the phone to an existing extension. Click “Assign Ext” You will be
prompted for the extension number.
2. Create a new extension for the phone. Click the “Add Ext” button. You will be
taken to the create extension page and prompted for Extension name and number.
Click “OK” to create the extension.
3. Reject the phone. If the phone does not look familiar to you, or it has not been
authorized for use with PortSIP PBX, you can “Reject” to delete the provisioning
request.
By MAC - for legacy phones

You can add new legacy phones that do not support plug and play, as follows:
1. Click “Add Phone” from the “Phones” tab.

2. Select extension this phone uses.
3. Now select the phone model.
4. Enter the MAC address of the device and click “OK”.
5. You will be taken to the provisioning page of that extension.
6. You can optionally configure other settings for the phone.
7. When done, click “Save” to add the phone to the extension.
8. Configure the DHCP server to serve the configuration URL, or configure it from the
phone web interface.
Provision Remote extensions by RPS

If you are adding phones that are installed remotely, i.e. PBX in cloud, you must:
1. Click “Add Phone” button from the “Phones” tab.

2. Select extension this phone uses.
3. Select the phone model.
4. Enter the MAC address of the device and click “OK”.
5. You can optionally configure other settings for the phone.
6. When done, click “OK” to add the phone to the extension.
7. Now the PortSIP PBX will write the provision link to the IP Phone RPS. Once the
IP phone starts, it will obtain the link from the RPS to download the profile from
PortSIP PBX. Note: currently PortSIP PBX only support the PRS for Yealink,
Fanvil, Htek. If your IP Phone is not provided by one of them, please follow the
step 8.
8. Copy the provisioning link and insert to your IP phone manually.
Accessing the Phone UI

PortSIP PBX allows you to easily access the password protected web interface of your
configured phones. PortSIP PBX will provision them with a username and unique password
and manage the credentials for you. To access the Phone UI:
• Select the phone and click on “Phone UI”.

• For most phones, you will be taken straight to the phone UI page.
• For some older phones you might be asked to enter the password for the phone.
In this case, click the “Password” button, for the password to be shown, and copy
paste the password configured for the phone in the phone authentication page.
Changing Phone Settings
Changes made to the phone configuration from the “General” tab of the “Extensions” section
or within the “Phone Provisioning” tab of the “Settings” section for certain extension, will take
effect within 24 hours. You can re-provision the phones to force them to pick up the new
configuration immediately. If you need to re-provision the phones, for example after you have
made configuration changes:
1. Select the phones that you wish to re-provision.

2. Click “Reprovision”.
3. If the phone needs a reboot, it will be done automatically.
5.4 Extension Management

This section explains how to create and configure extensions in PortSIP PBX. There are
multiple methods to create an extension.
● When provisioning a new phone, you could choose to create a new extension.
● Extensions can be manually created from the “Extensions”.
● Extensions can be imported from a .csv file.
● Create the extension by calling REST API.
To configure an extension, click on “Call Manager > Extensions” in the PortSIP PBX Web
Portal. Click on “Add” to create a new one, or select an existing extension and click the Edit
button to configure or manage the existing extension users. “Web Access Password” is used
by extension users to log into Web Portal.
General
In the section of “General”, you can enter the extension number, password, first name, last
name and the email address of the user. The extension number can be numerals or letters; the
extension number and password are required. A welcome email with information on the
extension created, as well as voicemail and missed call notifications (configurable) will be sent
to the specified email address.
The field “Web Access Password” is used for the extension to sign in Web Portal.
If the SMTP server is configured, once an extension is successfully created and the its email is
set, PortSIP PBX will send an email to the extension's email which includes the extension
information and PBX parameters such as PBX SIP Domain, PBX IP, and the QR code. User
can use the PortSIP UC App to scan the QR code to login to PBX without enter the details.
There is a QR code for this extension, you can use PortSIP App to scan the QR code to sign in
the PBX rather than entering the information manually.
In the “Direct Inbound Dialing (DID)” section, you can select a DID for the extension instead
of creating an inbound rule separately.
“Exceptions” - create exceptions by entering the “Caller ID”, selecting the time frame in
“Received During” and chose the action in “Action” to bypass the extension forward rules.
Note: The email that you entered should be unique, duplicated email is not allowed since
PortSIP PBX v12, user can use extension's email to sign in the PortSIP PBX Web Portal.
Voicemail
The “Voicemail” tab allows you to configure the extension’s voice mail preferences (including
the voicemail PIN number for authentication), enable/disable PIN Authentication, play Caller
ID, and enable PortSIP PBX to read out the Caller ID and the Date/Time on which the
message was received.
After the extensions created successfully, the “Greetings for Voice Mail” section allows you to
configure your voicemail greetings.
Click the ""Browse"" button to upload the new greeting file, and click the “Lock” icon to specify
it as greeting file.
Forwarding Rules
Each extension can have a set of call forwarding rules that define what PortSIP PBX should do
when the extension user is unable to answer an incoming call. This can be configured on the
basis of following:
• The user’s status

• The time
Each status requires a call-forwarding rule. For example, if the user is unable to take a call
whilst their status is “Available”, you can forward the call to voicemail (The voicemail must
enabled in the "voicemail" tab) or to the mobile phone number.
Note: forwarding the call to certain mobile number requires the VoIP provider and
outbound rule configured.
Options
The “Options” tab allows you to configure options, restrictions and access for the extension:
• Outbound Caller ID – Outbound Caller ID could be entered here for extension, so

that when the extension starts external calling via certain provider/SIP trunks, an
outbound caller ID could be a replacement for certain SIP field. For more details,
please refer to Section [5.6](#Configuring VoIP Provider / SIP Trunk).
• Caller ID for External Emergency Call – It could be entered here for extension, so
that when the extension starts emergency external calling via certain provider/SIP
trunks, this ID could be used as a replacement of certain SIP field. For more
details, please refer to Section [5.6](#Configuring VoIP Provider / SIP Trunk).
• Record audio calls – If checked, all calls for this extension will be recorded as wav
file.
• Record video calls – If checked, all video calls for this extension will be recorded
as AVI file.
• Enabled – If un-checked, the extension will be disabled.
• Allow Paging/Intercom – If checked, the extension will be allowed to make Paging/
Intercom calls. This options can't be changed since it is inherited from the
"Extension Groups".
• Allow External Calls – If checked, the extension will be allowed to make call to
external number via configured VoIP Provider/SIP Trunk. This options can't be
changed since it is inherited from the "Extension Groups".
• Allow Web Portal Access – If checked, the extension will have the access to PBX
Web Portal. This options can't be changed since it is inherited from the "Extension
Groups".
• Belonged groups - It indicates the extension groups that this extension belongs
with.
Office Hours
The Office Hours Scheduling feature allows a user’s status to be changed on the base of
global office hours or specific office hours.
Select if the extension would follow the Global Office Hours, or use Specific Office Hours. To
specify Specific Office Hours, enable the option and choose the time for a week, and click left
or right arrow to apply in use.
Phone Provisioning
The "Phone Provisioning" tab allows you to add or edit settings of phones linked to this
extension. The management of IP phone settings is discussed in “Phone Provisioning”.
BLF
You can configure the BLF lights on an IP Phone in this tab.
Match a BLF button with an extension, so that this button will show the status of that extension.
The number of available BLF buttons varies per phone.
The following options are available for BLFs:
• BLF – shows presence of another extension.

• Speed Dial – link to a phone number for easy calling
• Custom Speed Dial
• Change status
Billing
The admin/tenant can set the balance for extension. When billing is enabled and the balance is
not enough (see section [5.13](#5.13 Billing)), the call will fails.
Profile
You can configure the extensions profile here. The company name and company website
cannot be modified. These fields are inherited from administrator’s profile when the
administrator creates extensions.
5.5 Extension Groups

Extensions and administrators could be managed under “Extension Group” of Call Manager.
Extension groups are used to determine what and to whom the information is shown. The
defaulted extension group “DEFUALT” cannot be deleted or modified.
Note: that an extension has to be a part of at least one group. When a new extension is
created, it will be grouped into “DEFAULT” by automatically.
Users can be assigned permission to view details about other members in their group, and
managers can be assigned elevated rights over users in their group. Rights are assigned on
the basis of Group membership, which means that a manager will be able to see call details of
any member of their group, regardless of the call destination or origin.
Creating Extension Groups

On the left menu of Web Portal, select "Call Manager > Extension Groups", and click Add
button. Fill in the Group Name and Group Description in Group Information, and choose the
Group Member Rights to be set.
By clicking "Group Members" tab, you could add existing extension users into the group.
Once finished, click the Apply button to complete the creation of group.
Once an extension group is granted the permission “Allow Access to Web Portal”, all
extensions in this group could sign in PortSIP PBX Web Portal. Assume the "web access
password" for extension 101 is 111111, the SIP domain set in PBX system is portsip.io, and
the extension 101 belongs to default group which has been granted with login permission to
the system Web Portal, extension 101 could login with below info:
• Choose "Sign in with Extension Number"

• Extension number: 101
• Extension web password: the web password of extension 101, in this case it is
111111
• SIP Domain: portsip.io
An extension may be assigned to various group simultaneously, and owns a collection of the
permission for these groups.
Monitor
The extension group contains a "Monitor" tab, which is used for setting the monitor
permissions for the extensions. In this tab, you can choose multiple extensions to assign them
the "Monitor" permissions. The monitor members can silence monitor other extensions’ calls,
whisper, barge-in, and barge-break the extensions calls. It is commonly used for CTI purpose.
For more details please see [CTI](#11.3 CTI) section.
5.6 Providers / Trunks

VoIP providers/trunks “host” phone lines and replace the traditional telco lines. VoIP providers/
trunks can assign local numbers in one or more cities or countries and route these to your. In
most cases they also support number porting.
VoIP providers/Trunks are able to offer better call rates because they may have an
international network or have negotiated better rates. Therefore, using VoIP providers can
reduce call costs.
We recommend to use supported VoIP providers as all of our supported VoIP providers have
been tested for interoperability with PortSIP PBX, and are retested with each new build. The
configuration wizard allows you to quickly and easily add them.
PortSIP PBX supports two types of VoIP providers:
• Registration Based – These VoIP providers require the PBX to register with the
provider by using an authentication ID and password. Most of the VoIP providers
are predefined in PortSIP
• IP Based - IP Based VoIP Providers / SIP Trunks do not generally require the PBX
to register with the provider. The IP address of the PBX needs to be configured
with the provider, so that it knows where calls to your number should be routed
Only the admin can add/edit/update the provider/trunk. Once the admin adds a provider/trunk,
and the admin can specify the it available to one or more tenants.
The tenants can only see the trunks/providers which assigned to him by admin, and create the
inbound/outbound rule on its base.
Configuring VoIP Provider / SIP Trunk

First, you need to have an account with a VoIP service provider. PortSIP PBX supports most of
the popular SIP-based VoIP service providers/SIP Trunk, and we recommend to use one that
has been tested and approved by PortSIP as PortSIP PBX includes pre-configured templates
for these VoIP providers.
After you have created the VoIP provider account, you will need to configure the account in
PortSIP PBX. To do this:
1. Select “Call Manager > VoIP Providers/Trunks > Add”. Enter a friendly name for
this VoIP provider
2. Select the Country for the VoIP provider. If the country that the providers locates is
not listed, please choose “Generic”
3. Select your VoIP provider from the Provider drop-down list. If your provider is not
listed, select “Generic”.
4. The hostname of SIP server or IP may be prefilled. Compare these information
with the details that you have received from your VoIP provider and check if they
are correct. Depending on the VoIP provider that you are using, some fields will be
disabled, which means you do not need to change them
Note: For generic providers, you need to fill in relevant parameters for server by
yourself. Please consult your provide for more details
5. Transport. The transport which used for the PBX communicates with your provider
/ trunk, you should consult your provider and choose appropriate transport,
currently support UDP and TCP. The transport must added in PBX before add the
provider / trunk. For example, if your provider requires the TCP, you should add the
TCP transport in PBX, please refer to section [5.1](#5.1 Domain and Transport)
6. If your provider is verified on IP address and does not require registration, please
choose "IP Based" for"Authentication Mode"
7. If you have customized a provider such as the E1 gateway and it is located in the
same LAN with PBX, or other PBX/SIP servers, please check “Provider is
located in same LAN with PBX”
8. This provider only accepts single via SIP header. Usually the providers/trunks only
accept single via SIP header. It is selected by default
9. Enter the VoIP provider account details. Enter the Authentication ID/username and
password of your VoIP provider account. If your provider/trunk is IP based, no
need to enter them.
10. Associated IPs of Provider. For some providers/trunks, it maybe sends the INVITE
message to PBX from multiple IPs rather than from the host only. You need to
enter each IP here and click the "Add" button to add them
11. Click the "Tenants" tab, choose one or more tenants to allow them access this
trunk/provider.
12. Once a tenant is assigned with the trunk/provider, admin can also set the DID pool
for this tenant. When tenant creates the inbound rule based on this assigned trunk/
provider, he can only use the DID number from the DID pool.
13. Admin can leave the DID pool empty for a tenant, so that the tenant cannot create
the inbound rule based on this trunk/provider, but no matter to create the outbound
rule based on this trunk/provider.
14. The DID Pool format: allows to set the wildcard *. If a tenant is assigned with the
trunk/provider, and the DID pool is set as "*". When we assign this trunk/provider
to other tenants, we cannot set any DID pool for them (the DID pool can be left as
empty) since the "*" is already assigned to the first tenant.
15. The DID pool allows to set a numerical prefix and wildcard *. For example, setting
the DID pool as 44****** means all numbers started with 44 and a total length of 8.
Once the inbound rule is created, any number started with 44 and a total length of
8 can be set as DID number for this rule.
16. The DID pool allows to set a number range like 12000-18000, or 22000-22800.
Setting the number range requires the start number and end number use a same
prefix: 12000-22000 is not allowed. The numbers must has same amount of digits:
12000-180000 is not allowed. The start number must be less than end number:
18000-12000 is not allowed.
17. DID pool allows to set a semicolon-separated list of multiple ranges, for example:
123;1100-1200;44******
The PortSIP PBX will display all added providers/trunks status by clicking “Call Manager >
VoIP Providers/Trunks” menu.
Configure E1/T1 Gateway register to PortSIP PBX

Consider we deployed the PortSIP PBX on a cloud platform such as AWS, AZURE, GCE, and
wish to configure the E1/T1 gateway which located in local LAN as a trunk for the PortSIP
PBX, but the E1/T1 without static public IP, we can't configure the Authentication mode to "IP
Based" and "Register Based".
For this scenario, we can configure that E1/T1 is registering to the cloud PortSIP PBX from
local LAN, then the E1/T1 can act as the VoIP Provider / SIP Trunk works with PortSIP PBX for
make & receive calls.
Please follow the below steps to config the E1/T1 register to the cloud PortSIP PBX.
1. Select “Call Manager > VoIP Providers/Trunks > Add”. Enter a friendly name for
this VoIP provider.
2. Choose “Generic” for both provider country and provider brand.
3. Enter a domain for the "Host" filed, this domain doesn't require to exist, you can
enter any domain here, for example, portspitrunk1.io.
Important: ensure this domain does not equal to any tenant's SIP domain.
4. Please choose "Accept Register" for"Authentication Mode".
5. For the "Authorization Name" , you can enter any number here, for example
123456, the E1/T1 gateway will use this for the authorization when it registers to
PortSIP PBX.
6. For the password, you can enter any password here, the E1/T1 gateway will use
this for the authorization when it registers to PortSIP PBX.
7. Other settings are the same as the previous section for confining the "IP based"
and "Register based" VoIP provider/Trunk.
8. After applied the settings, now you can configure the E1/T1 gateway to let it
register to the could PortSIP PBX. In the E1/T1 settings, set up the step3 "Host"
as "SIP Domain/SIP Server", for example portsiptrunk1.io, set up the cloud
PBX public static IP as "Outbound Proxy Server"，set up the PortSIP PBX
transport port as the "Outbound Proxy Server port", set up the step 5, 6
Authorization name and Password as the username and password, then the
E1/T1 gateway can register to cloud PortSIP PBX.
9. You can sign in to the PortSIP PBX Web Portal to create the inbound & outbound
rule base on this E1/T1 gateway trunk.
Outbound parameters and Inbound parameters

After completing the setup for providers, you could also go to “Call Manager > VoIP
Providers/SIP Trunks” and click “...” button to choose "Edit" the Inbound/Outbound
Parameters for providers:
• In “Outbound Parameters” tab, you could set some rules to make changes for
headers of INVITE messages to be sent to VoIP providers/SIP trunks. For
example, “user” of “to” SIP header could be set to “Outbound Caller ID” of the
extension who starts the call. You can setup the “Outbound caller ID” of extension
in the “Options” tab of extension, see section 5.4.
• In “Inbound Parameters” tab, user could set rules to make changes to field values
of SIP messages for incoming calls.
NOTE: Both inbound and outbound parameters are advanced options. It’s recommended to
use default values.
5.7 Configuring Inbound/Outbound Rules

Outbound and inbound rules determines how PortSIP PBX routes calls on the base of certain
criteria. You can configure rules to control through which provider/Trunk a call will be placed,
for example, to route the calls through your VoIP provider on the basis of least cost routing.
You can also set DID (Direct Inward Dialing) numbers to allow to bypass the receptionist or
IVR and place calls directly to a user’s extension.
Creating Inbound Rules

Many companies provide users and/or departments with “Direct or DID numbers”, which
allow their contacts to bypass the receptionist and make calls directly. DID numbers is also
referred to as DDI numbers in the United Kingdom and MSN numbers in Germany.
Even if you make use of a virtual receptionist, a direct line/number is often preferable because
it’s more convenient for the caller.
Direct dial numbers are easily implemented by using “Inbound Rules”. DID numbers is
provided by your VoIP provider or Phone Company and are virtual numbers assigned to your
physical lines. Usually you are assigned a range of numbers. Please ask your Phone
Company or VoIP provider for more information about DID numbers.
You have to configure one VoIP provider/SIP Trunk before adding the inbound rules.
To add Inbound Rule:
• From the PortSIP PBX Web Portal, select "Call Manager > Inbound Rules >
Add"
• Enter a friendly name for the rule
• CID number mask: you can enter the CID number mask here, which the PBX will
use to identify the caller. You can add the number in it’s entirety, identifying a
single caller, or use the * as a wildcard. For example 0044********** will identify a
UK Caller and 004420******** will identify a caller from London. Note: the * digits
must matched number actually digits. If the number is 3 digits, then should use ***
The CID number mask also allow set a number range, for example:
00442012345670-00442012345680.
The CID number mask can be empty.
• In the "DID/DDI number/mask" field, enter the DID number as it will appear in the
SIP "to" header (The number your provider has been applied as your main, or first,
DID number). PortSIP PBX will match the number inserted into this field with the
"to" header, you can use a single * to match any DID number, or use a single
number likes 00442012345678;
The number range is disallow, and prefix and * is disallow - for example: 004420*
• Select which provider/SIP Trunk you wish to be associated with this DID and
inbound rule, only allow assign one provider with an inbound rule.
• Specify how you wish to forward incoming calls according to this inbound rule:
◦ End Call
◦ Connect to Extension
◦ Connect to Ring Group
◦ Connect to Virtual Receptionist
◦ Connect to Voice Mail of an extension
◦ Connect to Call Queue
◦ Forward call to external number
• You can specify that an incoming call should be forwarded differently if it is
received outside office hours
Note: you can create multiple inbound rules based on a same DID number, but all these
inbound rules should set the CID number mask, and the CID number mask must not be
conflicted.
Example: You have a DID number 442012345670. Now create two inbound rules: the CID for
the first rule is set to 0044**********, the DID number mask set to 00442012345670, and the
call is set to route to the call queue 8000; the CID for the second rule is set to 0033*********,
the DID number mask set to 00442012345670, and the call is set to route to the call queue
9000.
Now let all English-speaking employees to be agent of call queue 8000, let all French-speaking
employees to be agent of call queue 9000. When the caller calls to 00442012345670, callers
from UK will be routed to the call queue 8000 to talk with English agent, and callers from
France will be routed to the call queue 9000 to talk with French agent.
Office hours for Inbound Rules

In the "Office Hours" tab, you can set the office hours for the inbound rule so that the
incoming calls will be routed to different destinations on the basis of the current hour.
If "Use default Global Office Hours" is selected, the PBX will use the office hours specified
by admin/tenant;
If "Use specific Office Hours" is selected, customized office hours rules apply instead.
Exporting and Importing Inbound Rules

If you need to export your Inbound Rules to a .CSV file either for backup or to make any
updates, follow these steps:
1. Sign in the PortSIP PBX Web Portal

2. Click on the “Call Manager > Inbound Rules”
3. Click on the “Export” button to start exporting your inbound rules
4. Select a location and a file name for your exported inbound rule file and click
“Save”. Your rules will be exported and saved in the .CSV file
To create multiple inbound rules, insert necessary fields on a CSV file by using correct format,
and then import them back into PortSIP PBX by using the import function.
To import your inbound rules into PortSIP PBX from a CSV file:
1. Sign in the PortSIP PBX Web Portal.

2. Click on the “Call Manager > Inbound Rules", click the "Import”
3. Browse to the file that you want to import, select it and click “Open”
4. The rules will be imported into PortSIP PBX
Creating Outbound Rules

An outbound rule decides through which VoIP provider/Trunk an outbound call would be
placed.
The rule is decided by the user/extension who is making the call, the number that is being
dialed or the length of the number, or the extension group to which the caller belong.
To add outbound rules:
• From the PortSIP PBX Web Portal menu, navigate to "Call Manager > Outbound
Rules" and click "Add " button. Enter a name for the new rule in the prompted box.
• Specify the criteria that should be matched for this outbound rule to be triggered
with. In the “Apply this rule to below calls” section, specify any of the following
options:
◦ Calls to numbers started with prefix – Apply this rule to all calls
started with the number you specify. For example, enter “00” to specify
that all calls with numbers started with 00 should trigger this rule.
Callers should dial “00123456” to trigger this rule. You can specify
more than one prefixes, separated by “;”. For example, “00;123;88”
specifies prefixes 00 and 123 and 88. If the called number matches
one of these prefixes, this rule will be triggered.
◦ Calls from extension(s) – Select this option to define a particular
extension or extension range for which this rule applies. Specify one or
more extensions separated by semicolons, or specify a range by using
a “-”. For example 100-120.
◦ Calls to number with certain digits – Select this option to apply the
rule to numbers with a particular digit length, for example 8 digits. By
this method, you can capture calls to local area numbers or national
numbers without requiring a prefix.
◦ Calls from extension group(s) – Rather than specifying individual
extensions, you can select an extension group.
• Now specify how to match outbound calls with the criteria. In the "Make outbound
calls on" section, select up to three routes for the call. Each defined provider/trunk
will be listed as a possible route. If the first route is not available or busy, PortSIP
PBX will automatically try the second route.
• You can transform the number that matches the outbound rule before the call is
routed to the selected gateway or provider by using the "Strip Digits" and
"Prepend" fields:
◦ Strip digits – Allows you to remove one or more digits from the called
number. Use this option to remove the prefix before a call is dialed to
the gateway or provider if it is not required. For example the extension
make call to 002345, if you specify to remove two digits, the prefix “00”
will be removed before it is routed
◦ Prepend – Allows you to add one or more digits at the beginning of
the number if this is required by the provider or gateway. For example,
the extension make call to 002345, we specify 2 in the “Strip digits”
field and set “Prepend” to “+44”, the final called number which PBX
forward to VoIP provider/SIP Trunk will be +442345
Office hours for Outbound Rules

In the "Office Hours" tab, you can set the office hours for the outbound rule so that the
outgoing call will be routed to trunks or not depending on the current hour.
For example, if currently time is out of the specified office hours, the call fails even the
outbound rule is successfully matched.
If selected "Use default Global Office Hours", the PBX will use the office hours which set by
admin/tenant;
If selected "Use specific Office Hours" this outbound rule will use the customized office
hours.
5.8 Configuring Ring Groups / Paging / Intercom

The Ring Group feature adds powerful capabilities to your PortSIP PBX. Ring groups will help
you not to miss any important calls, whilst the Paging/Intercom feature allows you to make
announcements to groups of people rather like a PA system.
A ring group allows you to direct calls to a group of extensions. For example, you could define
a group of three sales, and have the general sales number "DID" ring on all three extensions at
the same time or one after the other. When you create a ring group, you assign it with a virtual
extension number. This will be the number used by the PortSIP PBX to "Address" to the ring
group.
To add a Ring Group:
• In the PortSIP PBX Web Portal, select "Call Manager > Ring Groups" and click
the "Add" button
• Now enter the ring group fields:
◦ Ring Group Number – This number identifies the ring group from
other extensions. Specify a new one as needed. Do not specify an
existing extension number
◦ Ring Group Name – Enter a friendly name for the ring group
◦ Outbound Caller ID - Once the outbound caller ID for the ring group
is set, when no members answer the call and forward the call to
external number on a provider / trunk, this outbound caller ID could be
a replacement for certain SIP field. For more details, please refer to
Section [5.6](#Outbound parameters and Inbound parameters).
◦ Ring Time – Specify how long the extension should ring for.
◦ Ring strategy – Select the appropriate ring strategy for this ring group:
▪ Ring Simultaneously: All Ring Group members will ring
at the same time.
▪ Prioritized Hunt: Ring each available member of the
group by specific order.
▪ Cyclic Hunt: Ring each available member of the group by
the sequence the members are added into the group. The
member who has not been rang from a call would take the
priority.
▪ Least Worked Hunt: Ring each available member of the
group by the order the members are added into the group.
The member who has not answered a call from this group
would take the priority.
▪ Paging/Intercom: This is a Paging or Intercom group (see
the next section for more details).
• In the section "Group Members", specify the extensions that should be part of this
ring group. Simply click on the extensions to add them to the ring group, and click
again to remove them from the group.
• In the section "Destination if no answer", you can define what should happen if
the call is not answered by the ring group
Paging
When creating the ring group, selecting the “Ring Strategy” with “Paging/intercom” would
allow someone to ring a group of extensions and make an announcement via the phone
speaker. The called party will not need to pick up the handset as the audio will be played via
the phones speaker. The person who’s paging will not hear any audio back from the people
being paged.
Intercom
When creating the ring group, selecting the “Ring Strategy” with “Paging/intercom” would
allow someone to ring a group of extensions and make an announcement via the phone
speaker. The called party will not need to pick up the handset as the audio will be played via
the phone speaker. The person paging will not hear any audio back from the people being
paged.
If the extension user wants to talk with the caller, he/she should press the “*” button to start
talking, and stop by pressing “#” button.
Important
Before using the Paging or Intercom feature, make sure you have specified the paging/
intercom prefix number by:
1. From the PortSIP PBX Web Portal, select “Advanced > Settings" menu, click
“Advanced” tab to add the paging prefix in the “Dial code” field (*11 for example)
2. Make sure that the user who is trying to page/intercom a group has the permission
to do so. If a certain extension user would like to start paging/intercom, select “Call
Manager > Extension Groups”, edit the group to which the extension belongs,
click “Group Member Rights” table, and check the “Allow Paging/Intercom”
option
There are two ways to commit Paging and Intercom:
1. Assume that you have created a ring group for which the group number 9000, and
selected the “Ring Strategy” with “Paging/intercom”. When dialing 9000, all
members of ring group 9000 will answer the call automatically and can hear from
caller but caller cannot hear back from members. If someone of the members wish
to talk with the caller, just press the “*”, and stop talking by press “#” key.
2. If extension 100 want to intercom with extension 101, just dial “**11101”, and
extension user 101 will answer the call automatically and talk with caller 100. In
this example, *11 is the value of “Dial Code”.
5.9 Configuring Virtual Receptionist/Auto-Attendant
The virtual receptionist feature allows PortSIP PBX to answer phone calls automatically. When
a call comes into the PortSIP, the caller is presented with a list of options. The caller can
choose the appropriate option by using the numbers on their phone keypad. You can
implement a menu by using this feature. A virtual receptionist is also known as an Auto
Attendant.
For example, "For sales, press 1. For support, press 2 or wait on the line to be
transferred to the operator".
You can configure various virtual receptionists, each of which owns a unique extension
number. Depending on your preferences, you may configure to answer calls on the base of
which line the call comes in and from, as well on whether the call is received inside or outside
office hours. For example, you can have a different prompt for outside office hours that does
not include the options to be transferred to groups/queues since there are not agents available
to take the calls.
Recording a Menu Prompt

Before create your virtual receptionist, you must decide the menu options you wish to offer the
caller and record the announcement. A sample would be, "Welcome to XYZ. For sales, press
1. For support, press 2 or stay on the line for an operator".
Note: It is recommended to put the number the user should press after the option, i.e. "For
sales, press 1", rather than “press 1 for sales". This is because the user will wait for the
desired option and then "register" what number to press.
Creating a Virtual Receptionist

You can create multiple digital receptionists and link them to a particular line.
To create a virtual receptionist:
• In the Web Portal menu, click “Call Manager > Virtual Receptionist", click "Add "
• Specify the name and extension number for the digital receptionist.
• By default, PBX uses the system-defined “Default.WAV” for prompt. Click on the
“Browse” button to select a file that you previously recorded for prompt menu. You
must save the file in WAV format in PCM, 16kHz/32kHz/48 kHz, 16 bit, Mono
format. (In Windows Sound Recorder you must use the "Save as" option to save
this format). Besides, user may also choose prompt language for virtual
receptionist in “Virtual Receptionist Language”
• The prompt when call is transferring - The prompt file which will be played when
the call is transferring after caller pressed DTMF.
• Virtual Receptionist Language - The language for the prompt files
• Menu options. Specify actions and the extension number or System extension
number for each of numeric keys. Default value is “No Actions Specified”,
referring that no specific actions will be taken in response to the key. If the action is
directed to specific extension, ring group, call queue or another virtual receptionist,
please also select the target extension number you desired
• User Input: this option allows you to determine when the auto attendant will begin
the search for an extension that matches the user’s input. The available options
are detailed below:
◦ When Extension Matches: The auto attendant will wait until the caller’s
digit sequence matches an existing account. Once the auto attendant
finds a match, it will call that extension. This mechanism is useful
when accounts of varying name length are used; however, it might be
annoying to callers who enter a non-existing number since the auto
attendant will never begin the search
◦ After 1/2/3/4/5 Digit Input: The auto attendant will wait until the correct
number of digits has been entered before it will begin looking for an
account that matches. If the account does not exist, the system will
play an announcement indicating that the extension does not exist.
◦ User Must Hit Pound: The auto attendant will wait until the user hits
the # sign before searching for an extension. This mode is useful in
variable-length scenarios
• Timeout allows you to specify how long the system should wait for an input. If it
receives no input, it will automatically perform this action. This is for callers who do
not understand the menu or who do not have a DTMF capable phone. When
ready, click “Apply” to save the virtual receptionist.
• If extension user enters a DTMF value or key that is not defined in step 4, the
action fails. User may define how the call should be handled in such case in
“Calling failed” section, and the extension number (if necessary)
Direct Destinations
The Direct Destinations feature is somewhat like a built-in version of the IVR system.
To direct inbound calls to specified extensions, you can use the pre-configured destination
fields and link them to pre-recorded announcements and user input options.
Using the sample shown below, the auto attendant’s welcome message will be as follows: “For
Sales, press 1. For Support, press 2. For Accounting, press 3. For all other inquiries,
press 0.” (The user input options are linked to extensions 555, 518, 511, and 570.)
When configuring straight forward, uncomplicated auto attendants, direct destination is a great
solution. However, when configuring auto attendants that require advanced IVR development
and functionality, the IVR node is recommended.
Once the direct destination links have been established, the system will call the destination
number whenever a caller enters the number that is associated with it. In the sample shown
above, when the caller presses 2, the call will be connected to extension 101.
By placing a pound sign after the direct destination (e.g., “2#”), the system will wait 3 seconds
before dialing the direct destination. This is useful if you have extension numbers in the 100
range (101, 102, etc.). The 3-second delay ensures that the caller’s complete input (e.g., 101)
will be processed rather than just the first digit.
• User Input: This number can be one or multiple digits; however, the system dials
direct destinations immediately after a user has provided keypad input, so
overlapping between a direct destination and an extension number can be a
problem. For example, extensions starting with “1” would conflict with a direct
destination of “1” because the system would be unable to dial the extension
number. The best way to avoid this situation is to choose extension numbers that
do not overlap with either direct destinations or mailbox and outbound call prefixes.
The extension range 4xx through 7xx meets these criteria. Wild cards can also be
used in this field.
◦ If circumstances render it difficult to change the extension assignments
(e.g., business cards with extension numbers already in circulation), a
timeout mechanism can be used. By placing a pound sign after the
direct destination (e.g., “1#”), the system will wait for 3 seconds before
dialing the destination.
◦ To redirect fax messages to a specific destination, you can use the
direct destination “F”. The CNG tone that announces a fax tone is
recognized by the system and is translated into the “F” key.
• Destination: This number can be either an internal number (e.g., an extension or
conference room) or an external number (must configure appropriate VoIP
provider and outbound rules)
Allowing Callers to Dial a Known Extension Directly

Whilst a digital receptionist prompt is playing, a caller can enter the extension number directly
to be connected to an extension immediately. This allows callers who know their party’s
extension to avoid going through a receptionist. This option is enabled by default. If you wish to
make use of this feature, simply instruct your callers by explaining this in the voice prompt.
For example, “Welcome to Company XYZ. If you know your party's extension number, you
may enter it now, otherwise, for sales press 1. For support press 2”.
Sending HTTP Request to 3rd Server Depending on User’s Input(Web

Hook)
When creating virtual receptionist, there are two tabs available for user: Virtual Receptionist
and Action URL. User may setup common Virtual Receptionist in “Virtual Receptionist” tab,
and define HTTP request and relevant actions in “Action URL”.
Action URL is applied as in below scenario:
When users call the Virtual Receptionist and dials the pre-configured DTMF key, Virtual
Receptionist will send a HTTP request as defined to the URL of a third-party server, and parse
the target extension number in respond message from the third-party server to forward the call
to the target extension.
• Name: Enter a user-friendly name for the HTTP request. This field is mandatory.
• Action Type: Choose the method to trigger Action URL. PortSIP PBX allows to
trigger the rule with user inputted DTMF key or caller number. Depending on his
request, user may choose “DTMF” or “Caller Number”. Once “DTMF” is chosen, if
the DTMF entered is replica to DTMF specified in “Virtual Receptionist” tab,
system will always invalidate settings in “Virtual Receptionist” and handle the call
as defined in “Action URL”.
• DTMF match list/ Caller number match list: Depending on the selection in
“Action Type”, user may specify the “DTMF match number” or “Caller number
match list”. User may enter a semicolon-separated list of numbers at one time,
e.g. “101;102;103”. The entered number must be unique and must not be
duplicated.
The match list also can specify to a numbers range, for example: 860000-880000,
it’s used for below scenario: someone calls to virtual receptionist and enters his
bank card number. If the number fall in the matched DTMF range, the virtual
receptionist will call the action URL to return some values to indicates the next
actions.
Once an item of the Action URL is triggered, an HTTP request will be sent to the
third-party server. User may specify the username and password for authentication
in “Credentials for HTTP Basic authentication with 3rd server” section (not
mandatory), and choose the method for sending HTTP request from POST or
GET. Fields “Connection timeout” and “Timeout for waiting response” are filled
to setup the timeout value for communication between Virtual Receptionist and
third-party server.
• Action (URL or number): Action to be executed will be entered here when the
preset action is triggered. If HTTP URL is entered here, Virtual Receptionist will
send an HTTP request to the third-party server and forward the call depending on
the returned value of HTTP request. If a DTMF number is entered here, Virtual
Receptionist will forward the call to the designated number
HTTP Request Message
PortSIP has defined below parameters to form up the HTTP request message to third-party
server in JSON format.
• "from": "var_caller_number" - Caller’s number, i.e. the caller number who’s calling
to Virtual Receptionist
• "to": "var_callee_number" - Callee’s number, i.e. the extension number for Virtual
Receptionist
• "input":"var_input_dtmf" - DTMF inputted by user
• "from_name": "var_caller_display_name" - Display name of caller. It will be left
empty if no value provided
• "account_name": "var_account_name" - Name of the Virtual Receptionist
Assuming that we had create a Virtual Receptionist with number 888 and named as Sales.
And its Action URL is defined as follows:
• Name: Action1
• Action Type: DTMF
• DTMF match list: 22, 33
• HTTP method: GET
• Action (URL or Number): http://www.appserver.com/dest.php (If a DTMF number is
filled here other than URL, Virtual Receptionist will forward the call to the extension
specified other than sending request to third-party server.)
When extension 101 (display name Jason) calls 888, Virtual Receptionist 888 will auto-answer
the call and play prompt to the caller. As extension 101 dials 22 or 33, Virtual Receptionist will
send below HTTP request in GET method: http://www.appserver.com/
dest.php?from=101&to=888&input=22&from_name=Jason&account_name=Sales
If POST is chosen for HTTP method, Virtual Receptionist will send below HTTP request in
JSON format by means of POST:
“from” : “101”,
“to” : “888”,
“input”: “22”,
“from_name” : “Jason”,
“account_name” : “Sales”
HTTP Response Message
PortSIP PBX has defined response to HTTP request sent by Virtual Receptionist as follows:
• "status_code": 200 or other possible status code, of which 200 represents

successful request and other refers to failure.
• “action”: Values including “call”, “hangup” and “repeat” indicates the action to be
taken by Virtual Receptionist.
◦ call – To forward the call to number as defined in “destination”
◦ hangup – To hang up the call directly
◦ repeat – To repeat the prompt message
• “destination”: The target callee number. It’s valid only if value for “action” is set as
“call”; otherwise it will be ignored.
“status_code” : 200,
“action” : “call”,
“destination” : “222”
Once Virtual Receptionist has received response as above, it will forward the call to extension
222.
5.10 Configuring Call Queue

Call Queue allows calls to be queued whilst agents (members of a call queue) answering calls.
Calls do not go unanswered but wait in a queue until an agent is available to take the call.
To add a Call Queue, select menu “Call Manager > Call Queue” and click “Add”. Now fill in
the necessary fields:
• Queue Number – Specify the queue number here. It should not be an existing
extension number
• Queue Name – Enter a friendly name for the Queue
• Outbound Caller ID - Once the outbound caller ID for the call queue is set, when
no members answer the call and forward the call to external number on a provider
/ trunk, this outbound caller ID could be a replacement for certain SIP field. For
more details, please refer to Section [5.6](#Outbound parameters and Inbound
parameters).
• Ring Duration – How long the caller would be queued
• Music on hold – The music that would be played when the caller is queued
• Skip member(s) who's calling - If this option was checked, the queue will don't
distribute calls to the agent who is on call
• Keep waiting if there is no members online - If this option was checked, even
there no any agent online, the queue will still keep the caller in the queue until
reached the maximum wait time
• Set the member ready to accept call automatically - If this option was checked,
once a queue agent(member) is registered to PBX, his status will be set to "sign
in the queue" automatically, then the queue will distribute calls to that agent, after
the agent was ringing or completed a call, the agent status will be set to "sign in
the queue" automatically.
If this option was unchecked, once the queue agent is registered to PBX, his
status will be set to "sign out the queue" automatically, the agent must use REST
API or dial a code to sign in the queue in order to let queue distribute the calls to
him. After the agent was ringing or completed a call, his status will be set to "sign
out the queue" automatically.
• Polling strategy – This option allows you to choose how calls should be
distributed to agents:
◦ Ring Simultaneous: All Ring Group members will be rang at the
same time.
◦ Prioritized Hunt: Ring each available member of the group in
configured order
◦ Cyclic Hunt: Ring each available member of the group by the order
the member was added. The member who has not been rang
previously will take the priority
◦ Least worked Hunt: Ring each available member of the group by the
order the member was added to the group. The member that hasn't
answered a call from this group takes priority
Configuring Queue Options

• In the “Destination if no answer” section, you can define what should happen if
the call does not get answered by an agent. If no agent logged into the queue, this
option would be triggered immediately
• In the “Other options” section, you can specify a custom introduction prompt and
a custom music on hold file. You can now choose whether to play the full intro
prompt before the system starts to call queue agents
• Maximum Queue Wait Time. Once the caller stayed in the call queue longer than
this duration, it will be hangup by PBX.
• SLA time. SLA refers to service level agreement. Once it’s set, you will get a
notification every time when a call stays in the queue longer than the specified
SLA time.
SLA is used to make sure that your callers are queuing no longer than the time
you have specified.
For example you declare that all calls within your organization are answered within
3 minutes, you need to set the SLA in the queue to 180 seconds. Once that time is
reached the queue manager will receive an alert notifying that a call has breached
the SLA.
Configuring Queue Agents (members)

By clicking the “Members” tab, you can select the agents for the call queue, the queue
manager will never receive calls from the queue.
Notifications
You can set one or more extensions as the queue manager(s) to receive the email notifications
if the call exceeds SLA time or is lost.
If an extension is set as queue manager, PBX will not assign the call to him/her, which means
the queue manager will not receive the call even he is a member of the call queue.
Note: The SMTP server and the email of queue manager must be set up in order to receive
the notifications.
Lost calls
You can query the lost calls of the queue in this tab, for example, if the agents are busy and
the caller hang up the call, then the call will be listed here.
Set agent status to ready or not ready by REST API

When an agent is registered to PBX, by default his status is ready, that mean the he is
available to receive calls. If agent away, he can use the REST API to set his status to "Not
ready" then queue will don't send call to him.
You can find the REST API details here: https://www.portsip.com/pbx-rest-api/v12.2/

call_queue.html#member_state_update
The REST API POST request body likes below:
{
"extension_number" : "8000",
"status" : false,
"extension_id" : 244385281237716992
}
The extension_id is the id of extension which be the queue agents(members).
The extension_number is the queue number.
Set agent status to ready or not ready by dial code

The queue agent can dial a code to set his status to ready or not ready.
For example, extension 102 is the member of queue 8000, 102 can dial *51*8000 to set his
status to ready(sign in the queue) and use *52*8000 to set the status to Not ready(sign out
the queue).
If extension 102 wants to set the status to ready(sign in the queue) for all queues, just dial
*51*, set the status to Not ready(sign out the queue) for all queues, dial *52*.
5.11 Configuring Conference
When the PBX is successfully installed, you can create a conference room by selecting the
menu “Call Manager > Conference” and click the “Add” button.
To create a conference:
• Select the menu “Call Manager > Conference”, and click “Add” button.
• Select your conference mode from the "Conference Mode" drop-down list.
• Enter a conference Room Extension number which will be dialed by the
conference Participants to join the conference. It should not be an existing
extension number.
• Enter the suitable Subject for the conference to remind participant the topic to be
discussed.
• Enter the PIN of the "Conference Room” if necessary. If the PIN was set, the
Participants must enter the PIN when joining the conference.
• Enter the Admin PIN for the host. When a user enters this PIN, he/she will be
identified as the conference admin to host the conference.
• Outbound Caller ID - Once the outbound caller ID for the conference room is set,
when inviting an external number to the conference, PBX will place the call to
external number on a provider / trunk, and this outbound caller ID could be a
replacement for certain SIP field. For more details, please refer to Section
[5.6](#Outbound parameters and Inbound parameters).
• Enter the maximum number of "Maximum Participants" field that limits the count
of members who join this conference.
• Specify the count of videos in “Grids for Video Conference”. Value 1, 2, 3, 4, 6, 9
supported.
• Set the bandwidth used during video conference in “Video Conference Bitrate”.
The value rages 128 kbps – 2048 Kbps. The higher the value is, the better the
video experience would be.
• Choose “Video Conference Frame Rate” with the rage 5 – 30. Higher value will
guarantee fluent video experience.
• Choose “Video Conference Resolution” from range of QCIF to 1080P. Higher
resolution leads to larger load to bandwidth.
• Choose the Prompt Language for the vocal notices which will be used when user
entering the conference.
• Click “Apply” button to confirm creating the conference room.
Each conference room supports up to 200 participants. It may vary dependent on the server
CPU, memory and bandwidth.
5.12 Managing Conference

Joining Conference
Once the conference room is created, inform the participants the conference number (“Room
Extension”). Assume that the user sets Room Extension 8008 as the conference number, the
user can join the conference by dialing 8008 from any SIP clients.
Inviting participants into conference room

You can also invite an extension or the mobile phone/landline phone user to join the
conference. Please see below section for more details.
Managing the conference room

Once the conference room is created, select the menu “Call Manager > Conference” to list
available conference rooms. You can either edit the conference room or delete it by click the
"..." icon to expand the menu.
• Manage: select the “Manage” menu to manage the conference room and
participants, see next section.
• Edit: select the “Edit” menu to change the conference room settings, such as the
Room PIN, Admin PIN, Maximum Participants.
• Delete: End and remove the Conference.
Managing the conference participants
Check a conference room in the conference list, and click the “Manage” icon to manage the
conference room participants.
• Invite participant: Click the “Invite” button to select an extension from extension
list, or enter the extension number directly. PBX will start a call to the specified
extension.
Once the call has been answered, the invited extension will be joint into the
conference automatically.
Mobile number or PSTN number could also be entered here to be invited into the
conference but must configured the appropriate trunk and outbound rules.
• Lock: Once the conference is locked, other users cannot dial into the conference
room.
• Record: Start or stop the conference recording
• Mute: When the room has been muted, all participants can't hear from each other.
• Refresh: Refresh the conference room information.
• Recording files: List recording files of the conference room. They could be
downloaded and saved in local.
• Mute: Click the "Mute" button by the end of a listed extension to mute the selected
participant.
• Set as main: Set the participant video as the main screen of video conference.
• Hang up: Kick out a participant from the conference room.
5.13 Billing
PortSIP PBX allows administrators/tenants to define customized calling rate. To do this, please
go to "Call Manager > Billing".
Parameters of billing rules

There are few parameters need to be entered when create the billing rule:
• FreeSeconds - in seconds
• ConnectFee - in monetary units
• PostCallSurcharge - in percents (0.01 means 1%)
• GracePeriod - in seconds
• Price' - in monetary units per minute
• PriceN - in monetary units per minute
• Interval' - in seconds
• IntervalN - in seconds
The simplest parameters are ConnectFee, which is fixed amount of money charged for each
successful call regardless of its duration; and PostCallSurcharge which is additional charge
applied. It is calculated on the percentage of the amount charged, that is if the call costs 1
dollar and the PostCallSurcharge is 0.01, the actual amount charged will be 1.01 dollar.
The following picture illustrates how the calls are charged. The process starts with comparing
value of the GracePeriod parameter with the duration of the call. The GracePeriod parameter
determines the minimum duration of the call that will be subject of charge. Calls with durations
of less than this value are not charged at all. GracePeriod value of 0 second and 1 second
provide almost the same behavior except that when the GracePeriod is 1 second, connected
calls with zero duration won't be charged with connection fee.
A ConnectFee is charged immediately upon connection, and all calls shorter than Interval’ will
be rounded to Interval1 seconds. FreeSeconds are granted after the Interval1, so this part of
the call is not charged, and calls shorter than (Interval1 + FreeSeconds) will be rounded to
Interval1 seconds. If call is longer than (Interval1 + FreeSeconds) remaining portion will be
rounded up to multiple IntervalIN seconds. After that, the PostCallSurcharge is applied to the
total amount charged.
The call illustrated in the figure will be charged using the following formula:
• Connect fee - monetary units

• Price 1 - monetary units
• Price N - monetary units
• Interval 1 - seconds
• Interval N - seconds
• Post call surcharge - fractional based on the setting in the Tariff/Destination Set
(E.g. for 10% fractional = 0.1)
Adding Billing Rate

To add billing rate, go to “Call Manager > Billing” and click “Add”, and fill in the fields below:
• Number Prefix: Enter the specific number prefix. Once specified, all calls related
to numbers started with this number will be applied with this rule.
• Type: Specify this rule for inbound or outbound calls.
Other fields are described in previous [section](#Parameters of billing rules).
Importing/Exporting Rate
PortSIP PBX allows to import rates in batch by using “Import” button on “Call Manager >
Billing” page. Once imported, all the imported rates will be listed on “Billing” page. If the rate
to be imported is replica to an existing rate on PBX, the import process will fail.
Besides, all rate info online could be exported by using “Export” button on “Billing” page.
Once completed, user will have a downloaded CSV file with all rates inclusive.
Note: Only CSV supported for both import and export features.
Chapter 6: Tenant Management

PortSIP PBX is designed as Multi-Tenant, which means one PortSIP PBX installation can work
for multiple enterprise (companies) by creating more than one tenants, and each tenant will be
able to have their own PBX system.
6.1 Creating tenant

To create a new tenant, select the left menu “Tenant” and click the “Add”.
When creating a tenant, you can specify the tenant profile details such as username,
password, sip domain and office hours. A tenant can modify his profile after signing into the
Web Portal.
You can also limit the resource the tenant uses by clicking the “Options” tab. The “Capability”
section under this tab allows you to set the maximum extensions, maximum concurrent calls,
maximum ring groups etc.
In the "General" tab allows to set below parameters:
• Username - The tenant username that is used to sign in the Web Portal. It should
be unique.
• Domain - The SIP domain for this tenant. It should be unique.
• Extension is not allowed to delete recordings - If it's selected, the extension
can't delete the recording files even if he signs in the PBX Web Portal.
• Enable this Tenant - If this option is deselected, this tenant will be disabled, and
all extensions of this tenant will no longer be valid.
• Allow concurrent logins - If this option is selected, the tenant can sign in Web
Portal from multiple devices. If deselected, once tenant signs in, the login in
another PC/mobile phone will be invalid.
• Allow display extension password in local - If selected, when editing extension
in the "Call Manager > Extension > General", the extension password will be
displayed.
• Allow extension to create temporary meeting - If selected, the extensions of
this tenant will be able to create temporary meetings via REST API.
In the "Options" tab, set below parameters:
• Country - The country for the tenant

• Timezone - The timezone for the tenant
• Currency - The currency for the tenant
• Enable extension to modify personal SIP password - If selected, the extension
will be able to modify his SIP password
• Enable extension audio recording - If selected, audio calls of all extensions will
be recorded
• Enable extension video recording - If selected, video call of all extensions will
be recorded
• Capability - Used to set the capabilities for the tenant
The “Storage” tab allows to adjust the storage quota for Recording files, Voice Mails and the
Call Reports:
• Disk quota (MB) - The maximum disk quota allowed for this tenant
• Recordings - Current disk usage of recordings for this tenant
• Voice Mails - Current disk usage of voice mails for this tenant
You can set the way of deleting old recording files and voice mails in "Auto cleaning" section.
6.2 Deactivating tenant
To deactivate an existing tenant, select the left menu “Tenant”, and all tenants will be listed.
Click the “...” button to expand the menu and choose "Edit" menu from the tenant that you
want to deactivate, uncheck the “Enable this tenant” box and click “Apply” button. The tenant
will be deactivated and all the extensions belongs to it would be deactivated as well.
If you want enable it again, check the “Enable this tenant” box.
6.3 Deleting tenant

To delete an existing tenant, select the left menu “Tenant”, and all tenants will be listed. Click
the “...” button to expand the menu and choose "Delete" menu from the tenant that you want to
delete.
Note: Once the tenant is deleted, all extensions of this tenant will be deleted as well.
6.4 Managing tenant

PortSIP allows administrator to manage tenant and its settings including extension users. To
do this, please go to Web Portal, navigate to “Tenant” menu, select a tenant to be managed
and click “...” button and choose "Manage". Now user may setup or modify the settings for the
tenant and manage its extensions.
Once completed, user may click avatar on the top right of page to display the menu and
choose “Switch to Administrator” to switch back to administrator account, without the need to
logout of tenant account and re-login to administrator account.
Chapter 7. Call Statistics

The Call Statistics feature allows you to view the call log, current active sessions, call
recordings and can be configured to send an email containing specific report statistics about
calls to and from PortSIP PBX. You can also receive these reports with .CSV format.
7.1 Call Sessions

By using " Call Statistics > Call Sessions" menu, you can quickly monitor all the current calls
and details on PortSIP PBX.
Hang up an established call by clicking "Hung up" button from a call session. Click the
"Refresh" button to update the calls status.
7.2 Call Recording

By using “Call Statistics > Call Recording” menu, you can quickly list all the recorded calls
and details on PortSIP PBX.
You can select the call recording and click the "..." button to expand the menu to play , or
download or delete it.
7.3 View Call Details

Select the left menu “Call Statistics > Call Details”. All call logs will be listed. Click the “Next”
button to see more.
Note: You can view more call details by double clicking the CDR.
7.4 Call Reports

Reports are generated and sent automatically via emails, so that report creation can be
executed with a low priority and will not interfere with the PortSIP PBX.
Note: To receive the exported call report, please make sure you have correctly configured
SMTP mail server. To setup, please go to the Step 4 of Setup Wizard or go to "Profile > Mail
Server".
Select the “Call Statistics > Call Reports” menu and click “Generate” to create a new call
report.
• Select the date range for call histories.

• Enter the email address the report will be sent to.
• Choose your preferred Report Format from the drop-down list. Default is .CSV.
• Select the filtering criteria by call status. Selecting the “Any” option will include both
answered or unanswered calls, whilst selecting the “Answered” will include the
answered calls only.
• Select the filtering criteria by call duration, then enter the call duration range (in
seconds). For example, enter 10 to “From”, and 20 to “End”, so that the call report
will include all call histories with the call duration between 10 and 20 seconds.
Click the “Apply” button, the call report will be sent to the specified email.
Chapter 8. WebRTC
PortSIP PBX has inbuilt WebRTC client which allows you make & receive calls in the browser,
support send IM, file, picture, picture clip, audio and video message, support conferencing and
sharing screen.
Before starting to use WebRTC Client, please ensure you have been read these sections
carefully:
• The [2.6 Avoid HTTPS Certificate Security Warnings](#2.6 Avoid HTTPS

Certificate Security Warnings).
• The [3.1 Deploying PortSIP PBX in LAN](#3.1 Deploying PortSIP PBX in LAN).
• The [5.1 Add TLS/WSS transport](#5.1 Add TLS/WSS transport).
• The [17.3 Solve the self-signed certificates warning](#17.3 Solve the self-signed
certificates warning).
If you used a trusted SSL certificates, please click left menu node "WebRTC", the browser will
opens WebRTC client in a new browser tab (Note: Chrome, new Edge, Firefox are
recommended), you will just need to enter the extension number and password to sign in.
Note: Chrome, new Edge, Firefox are recommended, and up to date.
If you used the self-signed SSL certificates, please follow below steps:
step 1: open https://192.168.0.16:5065/ in a browser, the browser will warning the connection
is unsafe, see below screenshot, please just click the "Proceed to 192.168.0.16(unsafe)".
(Note: the 5065 port is the WSS transport that you added in the PortSIP PBX, if you used
another port for the WSS transport, you should change the port here:
https://192.168.0.16:wssport/ )
Step 2: please click left menu node "WebRTC", the browser will opens WebRTC client in a
new browser tab, the browser will raises warning again, please just click the "Proceed to
192.168.0.16(unsafe)". then you will just need to enter the extension number and password to
sign in.
After signed in, you can use the WebRTC client to make & receive calls.
Chapter 9. Contacts
PortSIP includes Contacts (Phone Book) feature to manage contacts. It enables to import and
export contacts to the CSV file.
Contacts not only allows easy dialing of contacts, but also enables to match incoming call to
customer names, so that caller is shown with its customer name instead of the caller ID.
The Contacts is synchronized with the PortSIP UC client apps and connected IP phones.
By clicking "Contacts" menu, all the contacts in the phone book will be listed. You can click the
"..." button to expand the menu beside a certain contact:
• Manage - To add the contacts into the phone book

• Edit - To change the phone book settings
• Delete - To delete the phone book
Creating a new Phone book

By navigating to "Contacts" menu, and clicking the "Add" button, you can create a new Phone
book and assign it to the specified extension group, so that the extensions of the extension
group will be able to access this Phone book.
Importing & Exporting Phonebook Entries

You can import contact entries from a CSV file with the column headers on the first line, and
each contact entry on a single line with fields separated by a comma. You may find the
example CSV file at PortSIP Website.
After you have downloaded that sample CSV file, you can fill in it with your contact entries. To
import the company contact entries into the PortSIP PBX, sign in the Web Portal:
1. Navigate to “Contacts” menu and click “Import” button.

2. Browse your saved CSV file, select it and click “Open” to import your contact
entries into the phonebook.
To export entries from the PortSIP Contacts to a CSV file:
1. Navigate to the “Contacts” menu and select a phone book, and click “Export”
button.
2. Enter the target CSV file and click “Save” to export your phonebook entries.
When an IP Phone is successfully provisioned, the phone books which assigned to this
extension will be Synchronized to IP Phone automatically as "Remote Phone Book" for IP
Phone.
Chapter 10. Logs

The PBX Logs is an important troubleshooting tool that may help you identify the cause of
simple (and possibly complicated) issues.
10.1 Event Log

The Event Log will display the information when an event occurs, for example, the media
server is connected or disconnected, an transport is created.
10.2 Activity Log

The Activity Log will display the information for an operation which is completed or failed, for
example, a conference room is successfully created.
10.3 Log Files

You can download the log files of the PBX and analyze it locally.
Chapter 11. Advanced

After successful installation, the PortSIP PBX Configuration Wizard will guide the user go
through a series of settings that elicit basic configuration data. After completing basic
configuration with the Configuration Wizard, you can commit detailed configuration by using
"Advanced" menu in Web Portal of PortSIP PBX.
Important Note: only the administrator is allowed to access the “Settings” menu to change
the settings. Neither the tenant nor extension could change the settings.
11.1 Settings
General
You can change the general settings by selecting "Advanced > Setting" in PortSIP PBX Web
Portal.
Note: Usually we recommend NOT to change the default settings.
• Log Level: To output all SIP messages (sent and/or received) to log file in an
easy-to-read manner. The log file is named as "callmanager_log". Set the log level
as “None” will make significant improvement to PBX performance.
• Enable IPv6: This option could be used to enable or disable support on IPv6.
• Disable DIGEST authentication: If DIGEST authentication is disabled, the
authorization will be disabled as well. This option is deprecated (do not disable
DIGEST challenges).
• Disable auth-int DIGEST authentication: Once this option is checked, auth-int
quality of protection will be disabled.
• Disable authentication of mid-dialog requests: The PBX will not require
authentication of all requests in dialogs if this option is selected.
• Send 403 if a client sends a bad nonce: Send 403 if a client sends a bad nonce
in their credentials with this option selected. A new challenge will be sent if this
options is un-selected.
• Allow “to” tag in Registrations: Allow "to" tag in REGISTER message.
• Statistics Log Interval: Specify the interval for writes of the stack statistics to the
log files. The default value is 600 seconds.
• Enable Congestion Management: Use this option to enable/disable the
congestion management.
• Congestion Management Metric: The recommend value is WAIT_TIME. This
value is dependent on the expected wait time for each FIFO; this is calculated by
multiplying the size with the average service time.
• Congestion Management Tolerance: Congestion Management Tolerance for the
given metric. This determines when the Rejection Behavior changes. The default
value is 80.
Value Description
Percentage of max tolerance -> NORMAL (Not rejecting any

80
request);
80 - Percentage of max tolerance -> REJECTING_NEW_WORK

100 (Refuses new work, not continuation of old work.);
Percentage of max tolerance -> REJECTING_NON_ESSENTIAL

(Rejecting all work that is non-essential to the system (i.e. if dropping
> 100
something is liable to cause a leak, instability, or state-bloat, don't
drop it. Otherwise, reject it.).
• Automatically create the extension when a non-existent extension tries to

register: If this option is selected, when a non-existent extension registers to PBX,
the PBX will create this extension automatically. The default password for this new
extension is “portsip”.
• Enable PRACK (Reliability of Provisional Responses): If this option is selected,
the Reliability of Provisional Responses (RFC3262) will be enabled.
• Enable Flow Routing: To enable RFC5626.
• Close the session if no RTP packet received within specified period: The
PortSIP PBX tracks idle time for each of existing sessions (i.e. the time within
which there were no packets received), and automatically cleans up a session
whose idle time exceeded the value specified at compile time (120 seconds by
default).
• Enable the session timer (RFC4028): Enable the session timer (RFC4028) to
detect if the caller and callee are online. If this option is selected, the PBX will send
repeated INVITE requests to both caller and callee. The call will be hung up by
PBX if the INVITE is not correctly responded.
• Session timer duration: Specify the session timer duration during which the PBX
will send INVITE message to caller and callee. Default value is 120 seconds, and
the minimize value is 90 seconds.
• DNS Server: Specify the DNS server here, which overrides default OS detected
DNS server list. If it is left blank, the PortSIP PBX will use default DNS server for
system.
Port Range
You can change the Media server RTP range by selecting "Advanced > Port Range" in
PortSIP PBX Web Portal.
Advanced
You can change the advanced settings by selecting "Advanced > Settings" in PortSIP PBX
Web Portal.
• Dial code: Specify the prefix for making the Paging/Intercom call. With this prefix
specified, when the calling number is prefixed with dial code, the PBX will process
the call as Paging/Intercom. For more information, please see Section “Paging”
and “Intercom”.
• Alert-Info header for Auto Answer: Choose the “Alert-Info” header’s value,
which will be inserted into the SIP INVITE message when making Paging/Intercom
call.
For example, if “alert-autoanswer” is chosen, the below header will be inserted
into SIP INVITE message: “Alert-Info:info=alert-autoanswer”
Once the extension IP Phone detected “Alert-Info”, it will answer the call
automatically and turn on the speaker.
• Enable Call-Info header for Auto answer: Insert the “Call-Info” header into the SIP
INVITE message when making Paging/Intercom call.
For example, if this option is selected, the below header will be inserted into SIP
INVITE message:
“Call-Info: sip:portsip.com;answer-after=0”
Once the extension IP Phone detected “Call-Info”, it will answer the call
automatically and turn on the speaker.
• Require Answer Mode (RFC5373): Insert the “AnswerMode” into the SIP INVITE
message when making Paging/Intercom call.
For example, if this option is selected, the below header will be inserted into SIP
INVITE message:
“AnswerMode: auto”
Once the extension IP Phone detected “AutoAnswer” as “auto”, it will answer the
call automatically and turn on the speaker.
Different IP phones support different auto answer modes. Please refer to your IP
Phone manual to choose the correct mode.
• Busy Lamp Field: In this section, you could check “Enable Dialog State Agent”
and enter value for “Ringing Call Prefix” and “Held Call Prefix” respectively to
enable Busy Lamp Field feature for calls. Default value for “Ringing Call Prefix” is
, and ## for “Held Call Prefix”.
For example, an administrative assistant can see the status of their supervisor's line so he or
she knows when their boss is on the phone. Speed Dial is also available, which means the
assistant can pick up the phone, press the line configured to their supervisor's line, and their
supervisor's extension will ring.
If the boss’ extension is 101:
1. The assistant could use his/her IP phone to dial **101 to pick up the boss’
incoming call, if there no available incoming call, PBX will try to pickup the held call
2. The assistant could dial ##101 on his/her own IP phone to pick up the call which
held by boss
11.2 CTI
The PortSIP PBX support advanced CTI features, for example, silence monitoring, whisper,
barge-in, barge-break.
CTI Features
Silent call monitoring allows a supervisor to eavesdrop on a call conversation. The most
common scenario is in a call center where a call agent is speaking with a customer. Call
centers need to be able to guarantee the quality of customer service that an agent in a call
center provides. With silent monitoring, the supervisor can hear both call participants, but
neither of the call participants can hear the supervisor.
Silent monitoring is call based. When a supervisor invokes a silent monitoring session, the
following occurs:
• The supervisor selects a specific extension call to be monitored.

• The agent and customer voice will be sent to supervisor call automatically. The
monitoring call does not get presented to the agent and customer.
If you want someone (for example, the call queue manager) do silence monitor, you will follow
below steps:
1. Click the menu "Call manager > Extension groups", Selected an extension
group and click the "..." icon to pops up the menu, then choose "Edit", click the
"Monitor" tab, choose the extensions who you would like to grant the monitor
permission to him and checked the "Allow Monitor" the click "Apply" button.
2. Click the menu "Advanced > CTI", checked "Enable Monitor" option, by default,
the silence monitor service number(Monitor Number) is 888, you can change it if
you like.
Example:
We would like to let the extension 101 and 102 to silence monitor other extension's calls.
1. Click the menu "Call manager > Extension groups", create a new group which
named "Queue Manager"
2. Selected this extension group and click the "..." button, choose "Edit" menu, click
the "Monitor" tab
3. In "Monitor" tab, add the 101 and 102 into the monitor members
4. Checked the "Allow monitor" option and click the "Apply" button
5. Click the menu "Advanced > CTI", checked "Enable Monitor" option, and click
the "Apply" button.
6. Now if the extension 103 has a call, then 101 or 102 can silence monitor 103 by
dial "888*103"
7. During silence monitoring, the 101 or 102 can press DTMF 2 to whisper, press 3 to
barge-in, and press 4 to barge-break the 103's call
8. The 101 or 102 can dial "888*103*1" or "888*103" directly to silence monitoring; By
dial "888*103*2" to whisper 103, dial "888*103*3" directly to barge-in, and dial
"888*103*4" directly to barge-break the 103's call.
Exclusive Agent
In some scenarios, for the special industry callers of contact center, we will need the special
agents who have rich special industry knowledges and skills to serve them.
PortSIP PBX provides the "Exclusive Agent" feature allow set up one or more agents from
the queue as "Exclusive Agent" for the special callers, once the call comes from these
callers, the queue will distributes the call to the exclusive agent give highest priority if the
agent is idle, of course if all exclusive agents are busy / sign out, the call will be distributed to
other agents.
• Click the menu "CTI > Exclusive Agent", click the "Add" button
• Description: A descriptive name for the exclusive agent being entered. For
example, enter “XXX Bank” as the description for the bank caller.
• **Caller number:**enter the caller number who will be assigned the exclusive
agents. Once the call comes from this caller, the call will be distributed to the
exclusive agent with highest priority.
You can add more caller numbers by click the "Add" button.
• Call Queue: choose the queue member from the queues to set up as exclusive
agent
Note: if the call is not comes from the specified caller numbers, the exclusive agent
works as the normal agent.
VIP List
PortSIP PBX provides the VIP Caller feature, make VIP customers feel special when trying to
connect PortSIP PBX contact center. When the VIP call was determined, the queue always
give the top priority to the caller and pushed on top of the queue.
• Click the menu "CTI > VIP List", click the "Add" button
• Enter the VIP customer phone number
• Description: A descriptive name for the VIP Caller being entered. For example,
enter “Microsoft team” as the description for the VIP caller
• Set up how long of the the VIP number validity
• Enabled: turn on/off
Note: the VIP list is global validity for all queues
Harass Numbers
Spam calls are the plague of all businesses, especially call centers, PortSIP PBX provide three
ways for anti the spam calls.
1. A global "Number Blacklist", it will rejected the call silently if the caller is in the
number blacklist. You can find the details at section [12.2 Number Blacklist](#12.2
Number Blacklist).
2. Harass Number: PortSIP PBX also provides the "Harass Number" features for
anti the spam calls only for the Call Queue. The harass number is defined as two
levels, if a caller is determined in the "Level 1", the preset prompt file will be
played to alert the caller, and if caller press 1 the call will be hang-up, press 2 the
call is continue; If a caller is determined in the "Level 2", the preset prompt file will
be played to alert the caller, and the call will be hang-up automatically after play
finished.
Note: both of the two levels of the "Harass Number" only for the call which reached to
the queues, and its validity to all queues.
11.3 Managing Media Server

The media server is used for handling NAT scenarios and acts as a relay gateway for RTP
sessions of calls.
With the PortSIP PBX successfully installed, a built-in media server has been enabled by
default. The RTP packet from VoIP Endpoint A will be routed to Endpoint B with both IP and
Port translation during each call established.
Adding External Media Server

The PortSIP PBX uses default media server to relay RTP packets for calls. A large amount of
simultaneous calls will lead to high loads of CPU, network bandwidth, memory overload, voice
latency, unavailability for new calls, etc...
You can add more media servers to handle the RTP packets relay in order to reduce the
PortSIP PBX IP loads and decrease network latency.
Select the "Advanced > Media Server" menu in PortSIP PBX Web Portal, click “Add” and
enter a friendly name for the new Media server, and the IP of new Media Server (it could be
IPv4 or IPv6), and port number (default is 8896). Also please specify the maximum of call
sessions the media server could support on RTP data transportation.
Editing Media Server

You can view all the added media servers by clicking the menu “Advanced > Media Server”.
In the media server list, you can check the state for each server, such as enabled or disabled,
connected to PBX or disconnected. You may also configure the media server settings by
clicking “...” icon to popups the menu and choose the "Edit".
In the "Maximum call sessions" filed, you can specify the maximum call sessions the media
server could handle.
You can also disable a media server by turning off the “Enabled” switch button in the media
server list.
Removing Media Server

You can view the media servers by clicking "Advanced > Media Server". To remove a media
server, please click to select the server, and click the “Delete” button on the top of web page.
After a media server is removed, the PortSIP PBX will no longer use it to relay the RTP
packets.
Note: The Built-in(default) Media Server cannot be removed, but you can disable it by clicking
the “Enabled” button to disable it.
Be careful about the Built-in Media server. If you disabled it and did not add any other media
servers, the RTP packet will be sent directly between SIP endpoints during the calls, and if the
PortSIP PBX is running on internet, it may cause no audio and video transmit in the call.
11.4 Managing Conference Server
PortSIP PBX System provides multi-user conference features. Once the PBX successfully
installed, a built-in conference server is enabled by default. You can create as many
conferences as you like, as long as there still are free system resources (i.e. memory, CPU,
bandwidth) left.
Adding External Conference Server

PortSIP PBX uses conference server to handle the conference. The large amount of
simultaneous calls or a lot of conference servers will lead PBX server to high loads of CPU,
network bandwidth and memory, which eventually cause voice latency, and unavailability to
handle new calls.
You can add more conference servers to handle the conference in order to reduce the PBX
Server loads and decrease network latency.
In PortSIP PBX Web Portal, select the "Advanced > Conference Server" menu, click "Add"
button, and enter a friendly name for the new Conference Server, the IP of new Conference
Server (could be IPv4 or IPv6), and conference server port 8886. Also please enter the
maximum conference rooms and maximum participants, and click “Apply” button.
Editing Conference Server

You can view all the added conference servers by clicking “Advanced > Conference Server”.
In the conference server list, you can check the state for each server, such as enabled or
disabled, connected to PBX or disconnected. You may also configure the conference server
settings by clicking “Edit” icon button to popups the menu and choose "Edit".
In the "Maximum Rooms" filed, you can specify the maximum conference rooms for this
conference server that you can handle.
You can also disable a conference server by turning off the “Enabled” switch in the conference
server list.
Removing Conference Server

You can view all the Conference Servers by clicking "Advanced > Conference Server". To
remove a conference server, click to select the server to be removed and "Delete" button from
the top of webpage. Once the Conference Server is removed, the PortSIP PBX will no longer
use this Conference Server to handle conference.
Note: The Built-in Conference Server cannot be removed, but you can disable it by clicking the
"Enabled" switch.
Be careful about the Built-in Conference server. If you disabled it and did not add other
conference server, the conference feature will not be enabled.
11.6 Configuring Mobile PUSH

PortSIP PBX uses PUSH technology to wake up the smartphone when a call is received on
client. Mobile PUSH messages wake up PortSIP Softphone or other Client Apps on mobile
device so that a call or Instant Message can be accepted, reducing battery usage and
improving reliability.
Android phones receive PUSH notifications from Firebase Cloud Messaging Server; Apple
phones receive PUSH notifications from APNs.
PortSIP UC Client app has built-in push service enabled by default in PortSIP PBX. If you wish
to enable another app, you can create your own Firebase or APN account to support the push
notifications for your apps.
Configure PortSIP PBX for Mobile PUSH Notifications
1. Login to the PortSIP PBX Web Portal.

2. Navigate to “Advanced > Mobile PUSH > Add new APP” for setting up a new
app for receiving PUSH notifications.
3. Check “Enable” to enable PUSH notification.
4. Enter the App ID.
5. If necessary, enter Google Server Key and Google SenderID for Android clients,
and upload Apple Certificate File and Apple Private Key file for Apple clients.
6. Click “Apply” to apply the settings and restart all clients so they re-provision and
take the latest settings. The step by step guide for make the Mobile PUSH
notifications works with your app and PortSIP PBX, please refer to below topics:
1. Implement the PUSH notifications in Native iOS App with PortSIP PBX
2. Implement the PUSH notifications in Android App with PortSIP PBX
11.7 Configuring Voicemail

Set the extension number of voicemail
When the PortSIP PBX is successfully installed, the Voicemail service would be enabled by
default. You can specify the voicemail service extension number by clicking “Advanced >
Voice Mail” node in left menu.
Users could dial to read his voice mails. The default voice mail number is 999.
Set voice mail quota

PortSIP PBX allows you specify the disk quota to store the voice mails. The default value is
unlimited. You can also enter the number of days that they will be kept before they’re deleted
automatically.
11.8 Security
PortSIP PBX provides security features with main purpose to block any malicious attacks
targeted to the PortSIP PBX in case the admin has not taken necessary precautions at firewall
level. It works by detecting and blocking packet floods / DoS attacks or brute force dictionary
attacks within the scope of identifying and cracking the extension number and the password.
The Anti Hacking configuration page is accessible by clicking on the menu “Advanced >
Security”.
Detection Period
This is a time interval in seconds when counting starts but no action is enforced. To disable
security, set it to a higher value.
Failed Authentication Protection

This is the protection in case the attacker tries to use a dictionary attack to guess the password
set for a particular extension. To do this the attacker has to send numerous invites and after
the server sends a “Proxy authentication Required message” the attacker will send an invite
with authentication. With this feature, the attacker can only send 25 requests in an attempt to
crack the password. If an IP Address spams PortSIP with 10 wrong Authentication attempts in
“Detection Period”, that IP address will be blocked and put in the blacklist for the time specified
in the “SIP Blacklist time interval” parameter, by default 1 hour.
Failed Challenge Requests (407)

D.O.S attacks can send REGISTER/INVITE requests but do not reply to Challenge (407).
Configure the amount of "fake" requests that PortSIP PBX will accept per IP Address. If this
value is exceeded in "Detection Period" interval the source IP address is put in the Blacklist.
IP will remain blacklisted till "SIP Blacklist time interval" expires, by default 1 hour.
Level 2 security
This is the 2nd layer of protection. Here you can specify how many packets can be sent from a
unique source IP address. The default value is 2000 packets per second. If an IP Address is
sending more than 2000 packets per second, it means that there is something wrong. At this
point the attacker IP will be blocked until "Level 2 blacklist time interval" expires.
Level 1 security
This is the 1st layer in packets per second. If an IP sends more packets than the amount
specified per second, it will get blacklisted for the “Level 1 blacklist time interval”. Default
value is 5000 packets per second. At this layer, once that packet rate exceeds this layer, the
blacklist is enforced.
Once an IP address was blocked due to above rules, it will display in the 12.3 section, from
which you can add it into “Whitelist” manually.
11.9 Backup and Restore

PortSIP PBX has provided backup and restore feature to backup system settings and data,
which enables to easily restore system and data when necessary, or migrate the system from
one machine to another.
Backup
To backup the PBX, please:
1. Windows: copy the C:\ProgramData\PortSIP folder to backup it

2. Linux: copy the /var/lib/portsip folder to backup it
Migrate PBX to another Machine on Windows

To migrate the PortSIP PBX from one machine to another:
1. Backup the PBX as described in above steps, and download the backup folder to
local.
2. In the new machine which has the PortSIP PBX installed, copy that downloaded
backup folder to the c:\ProgramData\PortSIP\ folder of new PBX server (The
c:\ProgramData is hidden folder) to overwrite it.
Migrate PBX to another Machine on Linux

To migrate the PortSIP PBX from one machine to another:
1. Backup the PBX as described in above steps, and download the backup folder to
local.
2. In the new machine which has the PortSIP PBX installed, copy that downloaded
backup folder to the /var/lib/portsip folder of new PBX server to overwrite it.
11.10 License
Without a license, PortSIP PBX could work for up to 3 simultaneous calls and 10 extension
registrations. If you require more, you will need to purchase a license.
Feel free to contact sales@portsip.com to purchase the license.
Once you have received the license key, please click the menu “Advanced > License”, and
enter the key received.
PortSIP PBX requires Internet connection to verify the license key periodically. Please ensure
that your PBX server could be connected to Internet smoothly. If the license key verifications
fails, the PBX will be downgraded to free version which only allow maximum of 3 simultaneous
calls.
Do not let others know your license key. If PortSIP PBX detects a second user, it will be forced
into invalid and will downgrades to free version which only allow maximum of 3 simultaneous
calls and 10 extensions registrations.
Please contact PortSIP Support or reseller if you encountered any license key related issue.
Chapter 12. Blacklist and Codes
12.1 Codes and E164

PortSIP PBX allows to set the allowed country code and disallowed code in order to stop
extension dialing a specific country.
To allow or disallow the country code, please click the “Blacklist and Codes > Codes and
E164 > Allowed Country Codes”, you can select or de-select one of more than one country.
Disallowed Codes
User may use this feature to block the calls made on the trunks by specified prefix.
Go to "Blacklist and Codes > Codes and E164" menu, select the "Disallowed Codes" tab,
you can add the disallowed number prefix by click the "Add" button.
Number Processing
Select country: Pretty straight forward. Select the country you are in so the system knows
your country code. For our examples lets use US as our country.
Remove if same country: When this option is selected and you try calling the same country
using the E164 format the country code will be removed from the called number.
Example: Number dialed as +12345678910 will be converted to 2345678910.
International Dial Code: If you call a different country using the E164 format then the number
will be converted and the international dial code will be added. If you are in the US for example
this will be 011. This will enable you to make international calls without the "+".
Example: Number dialed as +44123456789 will be converted to 01144123456789.
Area Code: Here you can add your area code so if you are making calls within your area and
the area code is not required then you can strip it if the "Remove if same Area Code" is
selected.
Example: Our area code is set to 813. Number dialed as +181345678910 will be converted to
45678910.
National Code: If you need to add a national code to make calls then you can add a national
code here and it will be prepended to the number during the processing.
Example: National code set to 8. Number dialed as +12345678910 will be converted to
82345678910.
Add Prefix: A prefix can be added in case it is needed or if you want to use it to select an
outbound rule. E164 rules are processed before the outbound rules.
Example: Prefix is set to 2. Number dialed as +12345678910 will be converted to
22345678910.
12.2 Number Blacklist

PortSIP PBX allows you to block certain number. All requests associated with blacklist will be
blocked immediately.
To add the number into blacklist:

2. Click on “Blacklist and Codes > Number Blacklist” from the left menu
3. Click “Add” to add a new entry
4. Enter the number that you want to block and enter the description
12.3 IP Blacklist
PortSIP PBX allows you to whitelist and blacklist IP addresses. All traffic originated from
whitelisted IP addresses will be allowed through unchecked by the anti-hacking features. All
traffic originating from blacklisted IP addresses will be dropped immediately.
Adding a Whitelist Entry to PortSIP PBX

Assume that you have a remote office connected to your PortSIP PBX. Your remote office has
a public IP address of 123.123.123.123. Traffic from this IP address is trusted. To add this IP
address into whitelist, you’ll need to follow below steps:

2. Click on “Blacklist and Codes” > “IP Blacklist”.
3. Click “Add” to add an entry.
4. Enter the IP address that you want to allow – in this example it should be
123.123.123.123 (you can also enter the IP 123.123.123.0 and choose a Subnet
Mask to allow an IP range).
5. Choose “Allow” for “Action” field.
6. Add a description for the IP address, for example “My Remote office”.
7. Click “Apply”. An allow entry will be created in the IP Blacklist page for the
whitelisted IP address. All traffic originated from this IP address will not be checked
and the anti-hacking algorithms will not come into effect.
Blocking an IP Address or a range of IP Addresses
Let us look at another scenario. Assume that there is a distributed attack coming from the
following IP addresses – 41.202.160.2 and 41.202.191.5. These two IP addresses have
already been blacklisted by PortSIP PBX’s anti-hacking auto-detection mechanisms.
You would, however, want to blacklist all the range, since you are sure that you will never get
any traffic from these IP addresses. In this case, we will blacklist the whole range from
41.202.0.0 to 41.202.255.255, i.e. all the IP addresses that started with 41.202.

2. Click on “Blacklist and Codes > IP Blacklist”
3. Click “Add” to add an entry
4. In the “IP address” enter the first address of the network range you want to block.
For this example we will enter 41.202.0.0
5. Since we want to block all IP addresses started with 41.202, we will select a
Subnet Mask of 255.255.0.0. The range of IP addresses contained in this mask
will be displayed below
6. Set Action to “Permanently block“.
7. Enter a Description for this entry to help you remember why you added this entry,
for example “Anti D.O.S attack coming from 41.202.x.x”.
8. Click “Apply“. A block entry will be created in the IP Blacklist page. All traffic
coming from this IP address range will be checked, anti-hacking algorithms will
come into effect and all packets from this IP Address range will be completely
dropped and ignored
9. The PortSIP Blacklist / Whitelist mechanism does not conform a replacement of
firewall. It merely provides a defense mechanism to help differentiate traffic
trustable, and traffic not trustworthy. If, for example, you want to block all traffic to
your network and allow only your VoIP Provider IP address, you need to set this up
on your firewall.
When configuring a range of IP addresses in the Blacklist, you should also ensure that the
range does not include the IP address of which the PBX is installed.
Chapter 13. Profile

The admin and tenant user can manage their profile by selecting the “Profile” menu from the
PortSIP PBX Web Portal.
13.1 General
The admin user(admin also is a tenant) or tenant user can modify their profile details in the
“General” tab:
Username: The username for the admin or tenant user.
General Information
• Username: You can change the username (used for signing into Web Portal)
• Password: If the password was modified, the admin or tenant user must use new
password to login to Web Portal.
• Domain: The SIP domain for the tenant.
• Company name and company website: The company name and company
website for the admin or tenant user. The extension’s company name and
company website is inherited from the admin/tenant user who created the
extension.
• Email: The email for admin or tenant user, which is used for receiving notification
from PBX.
Capability
This section will display the capability of the admin/tenant. The tenant cannot modify the
capability for himself.
• Maximum Extensions
• Maximum simultaneous calls
• Maximum Ring Groups
• Maximum Call Queues
• Maximum Conference Rooms
• Maximum Virtual Receptionists
Options
• Country: The country of the tenant
• Time zone and Currency: The time zone and currency for the admin or tenant.
This setting will affect all extensions created by the admin or tenant.
• Billing for all outbound calls: If this option is selected, when the extension make
outbound call via trunk/VoIP provider, and the extension balance is not enough,
the call fails. And if the outbound call is established, but after a while, the
extension balance is not enough to make long call, the call will be hung up
automatically.
• Billing for all inbound calls: If this option is selected, when the extension
receives inbound call from trunk/VoIP provider, and the extension balance is not
enough, the call fails. And if the outbound call is established, but after a while, the
extension balance is not enough to make long call, the call will be hung up
automatically.
• Enable extension to modify personal SIP password: If this option is un-
selected, the extension can’t modify his SIP password.
• Enable extension audio recording: If selected, the extension calls will be
recorded as wav file
• Enable extension video recording: If selected, the extension video calls will be
recorded as AVI video file
• Record the call as dual tracks: If selected, the PBX will record the caller and
callee voice into dual tracks in the audio file, one track store the caller voice,
another track store the calee ovice.
• Extension is not allowed to delete recordings: If selected, the extension cannot
delete the recording files
• Allow concurrent logins: If this option is selected, the tenant can sign in Web
Portal from multiple devices. If deselected, once tenant signs in, the login in
another PC/mobile phone will be invalid
• Allow display extension password in local: If selected, when editing extension
in the "Call Manager > Extension > General", the extension password will be
displayed
• Allow extension to create temporary meeting: If selected, the extensions of this
tenant will be able to create temporary meetings via REST API
13.2 Office Hours

PortSIP PBX allows you to specify your office hours, after which the calls can be configured to
be routed on the base of the office hours. For example, in the office hours, calls will be routed
to your extension, and to voice mail when outside the office hours.
You can click "Configure" to set up the office hours for every weekday.
Holidays
You can also set up the holidays by clicking the "Add" button in "Holidays" Section. When a
call is made or received during the holiday, it will be treated according to the "Out of office
hours" rule.
13.3 Storage
The “Storage” tab allows to adjust the storage quota for Recording files, Voice Mails and the
Call Reports:
• Disk quota (MB) - The maximum disk quota allowed for this tenant
• Recordings - Current disk usage of recordings for this tenant
• Voice Mails - Current disk usage of voice mails for this tenant
13.4 Mail Server

To enable email notifications with PortSIP PBX, the SMTP details must be configured by going
to "Profile > Mail Server".
If you using the Google SMTP server, please make sure that you have “less secure” enabled
for your Gmail account. Please refer to below links for more details:
Less secure apps & your Google Account
You also need to select SSL or TLS security protocol if you’re using Google SMTP Server.
13.5 Music on Hold

“Music on Hold” could be leveraged to set the music on hold and the rules for playing.
• Enable: This box could be checked to enable “Music on Hold” so that when a call
is on hold, music will be played to the hold caller. To enable this feature, at least
one piece of music file must be specified.
• Personalized Music on Hold: Once this option is selected, music will be played
for the hold caller in random. Default playing mode is “Random music per day”.
• Random music per call: When Personalized Music on Hold is checked, user
could specify check this option so that music on hold for each call may differ.
• Random music per day: Default playing mode for Personalized Music on Hold.
• Music on Hold: Music could be uploaded here by clicking “...” button to navigate
the desired music files. To enable “Music on Hold” feature, this field is mandatory.
Only .WAV files supported currently.
◦ Music 1:
◦ ……
◦ Music 9: More music files could be uploaded by using these fields to
support on random music feature.
13.6 Event URL
Event URL
By setting up Event URL, PortSIP PBX is able to send CDR (Call Detail Report) and Extension
activity details to 3rd server by of HTTP request in POST method(Web Hook). The CDR is
formatted in JSON. To setup, please go to “Profile > Event URL” of Web Portal.
CDR Events
To send the CDR event to a 3rd server, below options should be provided:
Authentication method: The authentication method used when sending request to third-party
server. Both HTTP Basic Authentication and HTTP Digest Authentication are supported by
PortSIP PBX. If authentication is not necessary, please choose “None”.
• Username: Username used for authentication.

• Password: Password used for authentication.
• CDR URL: URL used for sending CDR to third-party server, e.g.
http://www.cdrserver.com/add.php.
Once set, CDR will be sent as below:
"answered_time":"1569567973",
"call_id":"b3dKobNlCwlOsjwyvlqD4A..",
"call_status":"ANSWERED",
"callee":"102",
"callee_domain":"portsip.io",
"caller":"101",
"caller_display_name":"",
"caller_domain":"portsip.io",
"did_cid":"",
"direction":"ext",
"ended_reason":"caller hangup",
"ended_time":"1569567981",
"fail_code":"0",
"final_dest":"sip:102@192.168.0.16:7059",
"outbound_caller_id":"",
"related_callid1":"",
"related_callid2":"",
"ring_duration":"5",
"ring_time":"1569567968",
"start_time":"1569567968",
"talk_duration":"8",
"tenant_id":"229806676134465536",
"tenant_name":"admin"}
Of the CDR messages sent, call_start_time, call_answered_time, call_ended_time,

target_add_time, target_answered_time and target_ended_time are all formatted in UNIX time,
which is a system for describing instants in time, defined as the number of seconds that have
elapsed since 00:00:00 Coordinated Universal Time (UTC), Thursday, 1 January 1970. User
needs to count the actual time with the timezone information.
Extension Events
To send the Extension events to 3rd server, below options should be provided:
Authentication method: The authentication method used when sending request to third-party
server. Both HTTP Basic Authentication and HTTP Digest Authentication are supported by
PortSIP PBX. If authentication is not necessary, please choose “None”.
Username: Username used for authentication.
Password: Password used for authentication.
Event URL: URL used for sending events to third-party server, e.g.
http://www.eventsserver.com/add.php.
Once set, the extension events will be sent as below:
“event_type”: “extension_registered”,
“tenant_id”: “admin”,
“extension_number”: “101”,
“source_ip”: “192.168.0.98”,
“time”: 1489482652,
“domain”: “sip.portsip.net”
13.8 Rebranding
The PortSIP PBX allows you customize and rebrand it.
Click the menu "Profile" and choose the "Rebranding" tab, in this page, it will allows you enter
the brand name, company name and upload logo file, you can simply have your own PBX.
Each Tenant can customize the PBX with his own brand name, logo.
If the tenant wish to display his own brand information, should open the Web Portal as this
way:
http://192.168.0.16:8888/login?tenantname=test
In above link, the 192.168.0.16 is the PBX IP, the parameter tenantname is use for specify the
tenant name, in case is for "test" tenant.
Chapter 14. REST API

PortSIP PBX offers the rich REST APIs. It allows you to implement your own web management
for the PBX rather than the PBX itself, and enables you to easily integrate the PortSIP PBX
with other system.
You can access the REST API user guide to learn more.
Note: The PortSIP PBX Web Portal is built based on REST APIs.
Calls related REST API

/api/extensions/hold - Allows use the REST API to hold a existing call by given extension
number or by given extension number and call session ID.
/api/extensions/refer - Allows use the REST API to refer the existing call by give extension
number or by given extension number and call session ID.
/api/extensions/attended_refer - Allows use the REST API to attended refer the existing call
by give extension number or by given extension number and call session ID.
/api/extensions/sessions - Allows use REST API to obtain all existing calls information by
given extension number.
/api/call_sessions/create - Initialize a call by given caller and callee number, the caller and
callee will receive the incoming call, after answered, the call will be connected.
Rewrite outbound caller ID

The outbound caller ID also known as CLI (Calling Line Identification), represents the phone
number of the calling (A) party in a phone call. It can be the full phone number or just a partial
representation (say, the last 4 digits) depending on the carrier, country and frame of mind of
the switch jockey.
PortSIP PBX allows set the outbound caller ID when make outbound call - just add a SIP
header into the INVITE message.
Example
If added X-Outbound-Cli SIP header to the INVITE message and make call to PBX, once the
PBX received call from client app, the PBX will rewrite INVITE message before send it to the
trunk.
X-Outbound-Cli: rewrite-from=123
Above header will cause the PBX rewrite the username of the From header to 123 when
sending INVITE to the trunk.
X-Outbound-Cli: rewrite-pai=456
Above header will cause the PBX add "P-Asserted-Identity" SIP header and set the
username to 456 when sending INVITE to the trunk.
X-Outbound-Cli: rewrite-rpi=789
Above header will cause the PBX add "Remote-Party-ID" SIP header and set the username to
789 when sending INVITE to the trunk.
If you would like to rewrite above headers only on specify trunk, you can also add below
parameter.
X-Outbound-Cli: rewrite-from=123;trunk-name:abc
Above header will cause the PBX rewrite the from header if the call is send to the trunk which
named "abc"; If no "trunk-name" parameter, the PBX will rewrite INVITE message on all
trunks.
Chapter 15. WebSocket Publisher

PortSIP PBX provides the Pub/Sub mechanism which bases on the WebSocket, the user is
able to create the WebSocket in any programming languages to subscribe to the PBX events,
once the subscribed events occur, PortSIP PBX will push the event message to subscriber
automatically, the message is in the JSON format.
PortSIP PBX provides below topics and keys for the Pub/Sub.
extension_events
All extension related event message will be published by extension_events topic，it includes
below message keys.
extension_register: extension registered to PBX or un-register from the PBX.

call_hold: call was held.
call_unhold: call has been resumed from held.
call_start: call starting.
call_established: the call was answered and successfully connected.
call_ended: call is ended.
call_noanswer: call is no answer.
call_reroute: the call was re-routed to another target.
call_fail: call is failed.
target_add: start call to a target. For example, extension 101 is registered to PBX from an IP Phone and an App, whe
target_ringing: the called target is ringing.
target_noanswer: there no answer from the called target.
target_fail: call failed from the called target. For example, the App / IP Phone rejected the call.
target_ended: call is ended from the called target. For example, the App / IP Phone hangs up the call.
cdr_events
Once a call is ended, the CDR of this call will be push to the subscribers, the message topic is:
cdr_events, the message key is below.
call_cdr: once a call is ended, the CDR will be packed in JSON format and push to subscriber.
queue_events
Once the queue status is changed, for example, the caller who in the queue was hangup the
call, or the caller who is in the queue is answered by an agent.... the related status information
will be pushed to the subscribers. The message topic is queue_events，includes below
message key.
queue_status: if the queue status changed, the information will be packed into JSON message and push to subscriber.
trunk_events
Once a trunk state is changed, for example, the PBX successfully registered to a Trunk, or
register to the Trunk is fails, or the registration is lost from a trunk, connection timeout, PortSIP
PBX will pushed the message information to the subscribers, the message topic is
trunk_events, the key is below.
trunk_connected
trunk_disconnected
More details of the WebSocket Pub/Sb, please read this link Going Real-Time with PortSIP
PBX Pub/Sub.
Chapter 16. Help

Skype ID: portsip, the name is "PortSIP PBX"
Email: support@portsip.com
Chapter 17. Best Practices for Deployment

Sometimes a bad production deployment can ruin all the efforts you invested in a development
process. This chapter aims to help you better understand how to deal with deployments in your
scenario and provide some best practices for deployments.
17.1 Deploying PortSIP PBX in LAN

This is a simple but typical deployment mode, in which scenario the PortSIP PBX is deployed
in LAN. Extensions from the same LAN will register to PBX and make calls to each other. With
default settings, the SIP signaling and RTP streams (RTP packet for audio and video) are
relayed by PBX.
In this deployment, when running the "Setup Wizard", in the step 1 you will need to enter the
"Private IPv4" only.
17.2 Deploying PortSIP PBX in LAN and registering to

public trunk
When the PBX is deployed in LAN, and yet it registers to a trunk that is located on Internet, the
caller can make call from landline/mobile phone to the extension of PBX, and the extension
can make call to a landline/mobile phone number.
This scenario requires the PBX server has a public static IP. When running the "Setup
Wizard", in the step 1 enter the "Private IPv4" and "Public IPv4". The public IPv4 is the static
public IP of the PBX server.
17.3 Solve the self-signed certificates warning

After you completed the PBX setting up, if get the self-signed certificates warning in browser
when you access PBX Web Portal by HTTPS or access the WebRTC client, please follow up
below steps to solve it.
1. Purchase a Domain from the domain provider for your PBX, for example Godaddy.
2. Add the an A record in the Domain DNS zone, resolve the Domain to your PBX IP.
3. Purchase a certificate from the trust certificate provider, for example Digicert,
Thawte, GeoTrust, usually you will have two files, one is certificate, another one is
private key. Note, please check the certificates for Nginx.
5. Click left menu "Home > Summary", then click the "Setup Wizard", in the step 1,
enter your domain for the "Web Domain", click Next button to complete the
wizard.
6. Click the left menu "Home > Call Manager > Domain and Transport", if there has
the TLS, WSS transport added, just delete the TLS and WSS transport.
7. Add the WSS and TLS transport with upload the purchased certificate files.
8. For Windows installation, restart the Windows Server directly.
9. For Linux installation, perform the below commands:

After restarted, you can sign in PortSIP PBX Web Portal by URL https://yourdomain.com:8887

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Open Source Software Used in this Product

Changes for Release v12.8.0

• Fix a potential crash bug of the WebRTC call

Changes for Release v12.7.2

• Fix a potential crash bug if the HA is enabled

Changes for Release v12.7.1

• Allows any extension obtain the contacts from PBX

Changes for Release v12.7.0

• Rewrite the push feature to improve the stability

Changes for Release v12.6.6

• OpenSSL was updated to 1.1.1n to avoid CVE-2022-0778

Changes for Release v12.6.5

• Support the tel scheme

Changes for Release v12.6.4

Changes for Release v12.6.3

• SSv2, v3, TLS 1.0, 1.1 have been disabled

Changes for Release v12.6.2

Changes for Release v12.6.1

• Removed the built-in backup/restore features, suggest use the VM snapshot to

b. Linux: by default it's

Chapter 1. Getting Started with PortSIP® PBX

1.1 What is PortSIP PBX

1.2 Before Started

A popular Linux distribution:

• CentOS 7.9 (64-bit)

Prerequisite knowledge for Windows

A Windows desktop or Windows server OS:

Cloud and Virtualization Environment Supported

• VMware ESX 5.X and above.

System performance depends on following key factors:

• Maximum simultaneous calls needed for PBX

• Easier access to Web Portal for PortSIP PBX

1.3 Getting Help and Support Resources

Chapter 2. Installation of PortSIP® PBX

2.1 Downloading PortSIP PBX

You will get the installer after download completed.

2.2 Installing PortSIP PBX on Linux

Preparing the Linux Host Machine for Installation

Installing a fresh PortSIP PBX v12.x for Linux

Configuring Linux Firewall Rules

2.3 Installing PortSIP PBX on Windows

Configuring Windows Firewall Rules

2.4 Upgrade the PortSIP PBX

2.6 Avoiding HTTPS Certificate Security Warnings

Assume your server IP is 172.217.14.16, you should open this URL:

Chapter 3. Architecture and deployment of

Running the PortSIP PBX Configuration Wizard

• Access the PBX Web Portal by visiting http://172.217.14.16:8888 or

3.1 Deploying PortSIP PBX in LAN

$ sudo docker exec -it portsip-pbx /bin/bash

Restart the Windows Server directly.

3.2 Deploying PortSIP PBX in the cloud

3.3 Deploying PortSIP PBX in other scenarios

In this page we can have an overview of the PBX statistics:

4.2 Services Status

4.4 System extensions

Chapter 5. Call Manager

5.1 Domain and Transport

Add UDP/TCP/WS transport

Add TLS/WSS transport

First of all, prepare the certificate files.